Session Initiation Protocol (SIP) for VoIP [Cisco IOS Software Releases 12.2 T]

Table Of Contents

Session Initiation Protocol (SIP) for VoIP

Feature Overview

Benefits

Restrictions

Related Features and Technologies

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Prerequisites

Configuration Tasks

Configuring the SIP User Agent (UA)

Changing the Configuration of the SIP User Agent (UA)

Configuring SIP Support for VoIP Dial Peers

Configuring a POTS Dial Peer

Configuring SIP Call Transfer for a POTS Dial Peer

Configuring SIP Call Transfer for a VoIP Dial Peer

Configuring Phone Number Translation Rules

Verifying the SIP Feature Configuration

Configuration Examples

Basic SIP Configuration Example

Configuring SIP with Multiple Codecs Example

Configuring Phone Number Translation Rules Examples

Call Transfer Configuration Examples

Command Reference

aaa username

debug ccsip all

debug ccsip calls

debug ccsip error

debug ccsip events

debug ccsip messages

debug ccsip states

default

gw-accounting

max-forwards

max-redirects

retry bye

retry cancel

retry invite

retry response

session protocol

session target (VoIP)

session transport

show sip-ua

sip-server

sip-ua

timers

transport

Glossary

Session Initiation Protocol (SIP) for VoIP

Document Update Alert

This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available.

If you are using Cisco IOS Release 12.3 or higher, refer to the following documentation in the Cisco IOS Voice Configuration Library, Release 12.3:

•Cisco IOS SIP Configuration Guide

If you are using Cisco IOS Release 12.2 or higher, refer to the following chapter in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2:

•Configuring Session Initiation Protocol for Voice over IP

Feature History

Release

Modification

12.1(1)T

SIP was introduced on Cisco Access platforms.

12.1(3)T

SIP Enhancements were implemented on Cisco 2600 series and Cisco 3600 series routers.

12.1(3)XI

The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was introduced and implemented on Cisco 2600 series, Cisco 3600 series routers, and the Cisco AS5300 universal access server.

12.1(5)T

The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was integrated into Cisco IOS Release 12.1(5)T.

12.2(2)T

SIP Enhancements were integrated into Cisco IOS release 12.2(2)T and implemented on the Cisco AS5400 universal gateway.

The SIP User Agent MIB feature was introduced and implemented on the Cisco 2600 series and Cisco 3600 series routers.

The SIP Diversion Header Implementation for Redirecting Number feature was introduced and implemented on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300 universal access servers.

12.2(2)XA

SIP and SIP Enhancements were integrated in Cisco IOS Release 12.2(2)XA and implemented on the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB

The SIP Gateway Support for Bind Command, SIP Gateway Support of RSVP and TEL URL, SIP INVITE Request with Malformed Via Header, Configurable PSTN Cause Code to SIP Response Mapping, RFC2782 Compliance for DNS SRV, SIP T.38 Fax Relay, and Call Transfer Capabilities Using the Refer Method features were introduced on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300 universal access server, Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

The SIP T.38 Fax Relay feature was implemented on the Cisco AS5300 universal access server, Cisco AS5350, and AS5400 universal gateways.

SIP, SIP Enhancements, and SIP Gateway Support of RSVP and TEL URL features were implemented on the Cisco AS5850 universal gateway.

12.2(2)XB2

The SIP Gateway Support for Bind Command, Configurable PSTN Cause Code to SIP Response Mapping, Call Transfer Capabilities Using the Refer Method, and SIP T.38 Fax Relay features were implemented on the Cisco AS5850 universal gateway.

12.2(4)XM

The ISDN Progress Indicator Support for SIP Using 183 Session Progress feature was implemented on Cisco 1700 series routers.

12.2(8)T

SIP, and the following SIP features were implemented on the Cisco 7200 series routers: SIP Enhancements, DTMF Relay for SIP Calls Using Named Telephone Events, SIP User Agent MIB, ISDN Progress Indicator Support for SIP Using 183 Session Progress, SIP Diversion Header Implementation for Redirecting Number, SIP Gateway Support of RSVP and TEL URL, SIP Intra-gateway Hairpinning, SIP INVITE Request with Malformed Via Header, Configurable PSTN Cause Code to SIP Response Mapping, RFC 2782 Compliance for DNS SRV, Call Transfer Capabilities Using Refer, and SIP T.38 Fax Relay features were integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers.

Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This feature was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

This document describes the Session Initiation Protocol (SIP) for VoIP on Cisco 7200 series routers in Cisco IOS Release 12.2(8)T and contains the following sections:

•Feature Overview

•Supported Platforms

•Supported Standards, MIBs, and RFCs

•Prerequisites

•Configuration Tasks

•Configuration Examples

•Command Reference

•Glossary

Feature Overview

Session Initiation Protocol (SIP)

Voice over Internet Protocol (VoIP) currently implements ITU's H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. Session Initiation Protocol (SIP) is a protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. The Cisco SIP functionality equips Cisco routers to signal the setup of voice and multimedia calls over IP networks. SIP provides an alternative to H.323 within the VoIP internetworking software.

The SIP feature also provides nonproprietary advantages in the areas of:

•Protocol extensibility

•System scalability

•Personal mobility services

•Interoperability with different vendors

The SIP feature includes the following functionality:

•Configurable in-band alerting

•Ability to specify the maximum number of SIP redirects

•Ability to specify SIP or H.323 on a dial-peer basis

•Support for both User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) transport layers for SIP messages

•Powerful debugging support

•Support for Domain Name System Server (DNS SRV) records for resolving SIP server host names

•Configurable SIP message timers and retries

SIP Enhancements

Beginning in Cisco IOS Release 12.1(3)T, the following enhancements to SIP were introduced:

•Configurable SIP message timers and retries

•Interoperability with unified call services (UCS)

•Support for a variety of signaling protocols, including ISDN, PRI, and CAS

•Support for a variety of interfaces, including

–Analog interfaces: FXS/FXO/E&M analog interfaces

–Digital interfaces: T1 CAS and E1 CAS

•Support for SIP redirection messages and interaction with SIP proxies. The gateway can redirect an unanswered call to another SIP gateway or SIP-enabled IP phone. In addition, the gateway supports proxy-routed calls.

•Interoperability with DNS servers including support for DNS SRV and "A" records to look up SIP URLs

•Support for SIP over TCP and UDP network protocols

•Support for Routing Table Protocol/RTP Control Protocol (RTP/RTCP) for media transport in VoIP networks

•Support for the following codecs (see Table 1):

Table 1 SIP-Supported Codecs

Codec

SDP

G711ulaw

0

G711alaw

8

G723r63

4

G726r16

2

G728

15

G729r8

18

•Support for Record-Route headers

•Support for IP Quality of Service (QoS) and IP precedence

•Support for IP Security (IPSec) for SIP signalling messages

•Authentication, Authorization, and Accounting (AAA) support. For accounting, the gateway device generates call data record (CDR) accounting records for export. For authentication, the SIP Gateway sends validate requests to the AAA server. For authorization, the existing access lists are used.

•Support for configurable expiration time for SIP INVITEs and maximum number of proxies or redirect servers that can forward a SIP request

•Expanded support for the mapping of Public Switched Telephone Network (PSTN) cause codes to SIP events

•Ability to hide the calling party's identity based on the setting of the ISDN presentation indicator

For more information, see the Configuring SIP for VoIP part in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2.

Call Transfer Capabilities Using the Refer Method

The Refer method provides call transfer capabilities to supplement the Bye and Also methods already implemented on Cisco IOS Session Initiation Protocol (SIP) gateways.

Call transfer allows a wide variety of decentralized multiparty call operations. These decentralized call operations form the basis for third-party call control, and thus are important features for Voice over IP (VoIP) and SIP. Call transfer is also critical for conference calling, where calls can transition smoothly between multiple point-to-point links and IP-level multicasting.

For more information, see the document Call Transfer Capabilities Using the Refer Method.

Configurable PSTN Cause Code to SIP Response Mapping

This feature allows customization of the standard RFC 2543 mappings between the Session Initiation Protocol (SIP) network and the Public Switched Telephone Network (PSTN).

For calls to be established between a SIP network and a PSTN, the two networks must be able to interoperate. One aspect of their interoperation is the mapping of PSTN cause codes, which indicate reasons for PSTN call failure or completion, to SIP status codes or events. The opposite is also true: SIP status codes or events are mapped to PSTN cause codes. Event mapping tables found in this document show the standard or default mappings between SIP and PSTN.

However, you may want to customize the SIP user agent software to override the default mappings between the SIP network and the PSTN. The Configurable PSTN Cause Code to SIP Response Mapping feature allows you to configure specific map settings between the PSTN and SIP networks. Thus, any SIP status code can be mapped to any PSTN cause code, or vice versa.

When set, these settings can be stored in the NVRAM and are restored automatically on bootup.

For more information about this feature, including configuration tasks and examples, see the document Configurable PSTN Cause Code to SIP Response Mapping.

DTMF Relay for SIP Calls Using Named Telephone Events

The DTMF Relay for SIP calls Using Named Telephone Events (NTE) feature adds support for relaying DTMF tones and hookflash events in SIP on Cisco VoIP gateways.

Note The DTMF Relay for SIP Calls Using Named Telephone Events feature is implemented for SIP only.

Using NTE to relay dual tone multifrequency (DTMF) tones provides a standardized means of transporting DTMF tones in RTP packets according to section 3 of RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, developed by the Internet Engineering Task Force (IETF) Audio/Video Transport (AVT) working group. RFC 2833 defines formats of NTE RTP packets used to transport DTMF digits, hookflash, and other telephony events between two peer endpoints.

DTMF tones are generated when a button on a touch-tone phone is pressed. When the tone is generated, it is compressed, transported to the other party, and decompressed. If a low-bandwidth codec, such as a G.729 or G.723 is used without a DTMF relay method, the tone may be distorted during compression and decompression.

The DTMF Relay for SIP Calls Using NTE feature adds SIP functionality. SIP IP phones currently do not have the capability to generate DTMF tones. Currently, DTMF tones are transferred using Cisco Proprietary RTP or transparently in band. The DTMF Relay using NTE feature allows SIP phones calling voice mail or other interactive voice response (IVR) systems to relay DTMF tones. Additionally, this feature prevents distortion of DTMF tones if the RTP session uses a low bit-rate codec, because tones are passed in NTE packets and are not compressed using the default codec.

With the DTMF Relay Using NTE feature, the endpoints can perform per-call negotiation of the DTMF relay method. During call setup, the calling and called parties negotiate to choose the DTMF relay mode. They also negotiate to determine the payload type value for the NTE RTP packets.

In a SIP call, the gateway forms a session description protocol (SDP) message that indicates:

•If NTP will be used

•Which events will be sent using NTE

•NTE payload type value

The DTMF Relay Using NTE feature also provides hookflash support using in-band and out-of-band modes. In in-band mode, the gateway relays the hookflash without notifying the application, and the default session application and any IVR scripts do not receive the hookflash.

In out-of-band mode, the gateway reports the hookflash to the application and the application can relay the hookflash to the next call leg.

Note In addition, the DTMF Relay for SIP Calls Using NTE feature does not support hookflash generation for advanced features such as call waiting and conferencing.

For more information, see the document DTMF Relay Using Named Telephone Event.

ISDN Progress Indicator Support for SIP Using 183 Session Progress

This feature provides support for handling inband treatments, such as call progress tones and announcements, when SIP is the session protocol for establishing call connections. The feature ensures the correct establishment of the media stream through the SIP network to allow the successful transport of in-band treatments, which might ingress from a PSTN node on a SIP gateway or egress to a PSTN node. The feature also allows VoIP calls using SIP to provide inband call treatment such as ringback tones, announcements when interworking with ISDN and channel associated signaling (CAS) PSTN networks.

SIP 183 Session Progress messages facilitate better call treatment for SIP VoIP calls when interworking with PSTN networks. The introduction of the 183 Session Progress message allows a called user agent to suppress local alerting from the calling user agent, and to play a tone or announcement during a preliminary call session, before the full SIP session is set up. This functionality enables the calling party to be notified of the status of the call without being charged for the preliminary portion of the call. A new Session header in the 183 Session Progress message controls whether or not the called user agent plays a tone or announcement for the calling party. The 183 Session Progress message is supported by default and does not require any special configuration.

RFC 2782 Compliance for DNS SRV

SIP on Cisco's VoIP gateways uses DNS SRV query to determine the IP address of the SIP Proxy or the Redirect Server. The query string generated has a prefix in the form of "protocol.transport." and is attached to the Fully Qualified Domain Name (FQDN) of the next hop SIP server. This prefix style from RFC 2052 has always been available; however, with this release a second style is also available. The second style is in compliance with RFC 2782, and prepends the protocol label with an underscore "_"; as in "_protocol._transport.". The addition of the underscore reduces the risk of the same name being used for unrelated purposes.

Use the srv version command to configure the DNS SRV feature.

For more information, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Diversion Header Implementation for Redirecting Number

The SIP Diversion Header Implementation for Redirecting Number feature provides support for a new SIP header field; Call Control (CC)-Diversion. The CC-Diversion header field enables the SIP gateway to pass call control redirecting information during the call setup. Call control redirection is the redirection of a call based on a subscriber service such as call forwarding. Call redirection information is information typically used for Unified Messaging and voice mail services to identify the recipient of a message. Call control rediversion information can also be used to support applications such as automatic call distribution, and enhanced telephony features such as Do Not Disturb and Caller ID.

If generated by the SIP gateway during call process, the CC-Diversion header field is based on the contents of the Redirecting Number Information Element (IE) in the ISDN Setup message. In addition, information such as the reason the call was redirected is included in the CC-Diversion header field.

For more information, see the document SIP Diversion Header Implementation for Redirecting Number.

SIP Gateway Support for Bind Command

Currently, Session Initial Protocol (SIP) signaling and media paths use an IP address that is provided by the IP layer as the source address. However, with the addition of the bind command, you can now configure the source IP address of signaling packets, or both signaling and media packets.

In previous releases of Cisco IOS software, the source address of a packet going out of the gateway was never deterministic. That is, the session protocols and VoIP layers always depended on the IP layer to give the best local address. The best local address was then used as the source address (the address showing where the SIP request came from) for signaling and media packets. Using this nondeterministic address occasionally caused confusion for firewall applications, as a firewall could not be configured with an exact address and would take action on several different source address packets.

However, the bind interface command allows you to configure the source IP address of signaling and media packets to a specific interface's IP address. Thus, the address that goes out on the packet is bound to the IP address of the interface specified with the bind command. Packets that are not destined to the bound address are discarded.

When you do not want to specify a bind address, or if the interface is down, the IP layer still provides the best local address.

The bind command performs different functions based on the state of the interface.

For more information, see the document SIP Gateway Support for the Bind Command.

SIP Gateway Support of RSVP and TEL URL

The SIP Gateway Support of RSVP and TEL URL feature provides the following SIP enhancements:

•RSVP

•Telephone URL format in SIP messages

•Interaction with forking proxies

•SIP intra-gateway hairpinning

•Reliability of SIP provisional responses

•Configurable screening indicator

•RFC 2782 Compliance (style of DNS SRV queries)

For more information, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Intra-Gateway Hairpinning

SIP hairpinning is a call routing capability in which an incoming call on a specific gateway is signaled through the IP network and back out the same gateway. This can be a Public Switched Telephone Network (PSTN) call routed into the IP network and back out to the PSTN over the same gateway, as shown below:

Similarly, SIP hairpinning can be a call signaled from a line (for example, a telephone line) to the IP network and back out to a line on the same access gateway:

With SIP hairpinning, unique gateways for ingress and egress are no longer necessary.

For more information about the SIP Intra-Gateway Hairpinning feature, including configuration tasks and examples, see the document SIP Gateway Support of RSVP and TEL URL.

SIP INVITE Request with Malformed Via Header

In the past, when an INVITE contained a malformed Via header, the gateway would print a debug message and discard the INVITE without incrementing a counter. However, the printed debug message was often inadequate, and it was difficult to detect that messages were being discarded.

Note This feature applies to messages arriving on UDP, because the Via header is not used to respond to messages arriving on TCP.

For more information about this feature, see the document SIP INVITE Request with Malformed Via Header.

SIP T.38 Fax Relay

The SIP T.38 Fax Relay feature adds standards-based fax support to SIP and conforms to ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks. The ITU-T standard specifies real-time transmission of faxes between two regular fax terminals over an IP network.

Much like a voice call, SIP T.38 Fax Relay requires call establishment, data transmission, and release signaling. The following figure shows the basic setup of SIP T.38 Fax Relay:

For more information, including configuration tasks and examples, see the document SIP T.38 Fax Relay.

SIP User Agent MIB

The SIP User Agent MIB addresses the need for SIP-specific gateway information to be made available by Simple Network Management Protocol (SNMP). The implementation of this capability is based upon the current IETF draft "draft-ietf-sip-mib-01.txt".

The implementation of the SIP MIB in the Cisco SIP gateway supports configuration objects related to SIP such as the configured SIP server, SIP timers, and number of retry attempts allowed for requests and responses.

The SIP MIB also supports SIP-specific statistical information objects. This includes information on numbers of provisional responses, success responses, redirection responses, client error responses, server error responses, and global error responses. In addition, the SIP MIB includes information regarding SIP Requests initiated and received as well as information about retries associated with each SIP Request type.

Benefits

Session Initiation Protocol

The SIP feature meets the needs of service providers that use SIP on the gateways of their VoIP network to:

•Enable Cisco voice-enabled platforms to provide RFC 2543-compliant user-agent client gateways

•Support codecs capable of Carrier-class voice quality

Although SIP is simpler than H.323, SIP provides similar capabilities in:

•System scalability

•End-to-end solutions

•High-density voice gateways

SIP Enhancements

The SIP feature enhancements enable SIP gateways to:

•Enable Cisco voice-enabled platforms to provide RFC 2543-compliant user-agent client gateways

•Support proxy-routed calls

•Redirect an unanswered call to another SIP gateway or SIP-enabled IP phone

•Allow end users to place calls on hold

•Hide the calling party's identity based on the setting of the ISDN presentation indicator

Call Transfer Capabilities Using the Refer Method

•SIP Call Transfer Using the Refer Method supports attended transfer and blind transfer in accordance with emerging SIP standards.

Configurable PSTN Cause Code to SIP Response Mapping

•The Configurable PSTN Cause Code to SIP Response Mapping feature offers control and flexibility. By using command-line interface commands, you can easily customize the default or standard mappings that are currently available between PSTN and SIP networks. This allows for flexibility when setting up deployment sites.

DTMF Relay for SIP Calls Using Named Telephone Events

•DTMF relay support for SIP

•Hookflash relay support for SIP

•Simultaneous support with Cisco Proprietary RTP (used for modem passthrough and modem relay)

•Provisioning of RTP payload type values

•Per-call negotiation of relay method and payload type values

•More accurate tone delivery

•Interoperability with SIP applications from other vendors

ISDN Progress Indicator Support for SIP Using 183 Session Progress

•Ensures that in-band treatments initiated in the PSTN are successfully transported through the SIP network

•Allows for internetworking of features between the PSTN and the SIP network so that the correct inband feedback is provided to the feature user

RFC 2782 Compliance for DNS SRV

•Compliance with RFC 2782 brings DNS compatibility. RFC 2782 updates RFC 2052 by prepending the protocol label with an underscore "_". This change reduces the risk of the same name being used for unrelated purposes. However, backward compatibility is available, allowing newer versions of IOS software to work with older networks that only support RFC 2052.

•Currently you must know the exact address of a server to contact it. SRV records enable administrators to use several servers to provide the same service within a single domain. SRV Resource Records (RRs) allow administrators to define primary and backup servers and move services from host-to-host without affecting service.

SIP Diversion Header Implementation for Redirecting Number

•Provides support for the Call Control (CC)-Diversion SIP header field

•Enables the SIP gateway to pass call control redirecting information during the call setup

•Redirection of a call based on a subscriber service such as call forwarding

•Unified Messaging and voice mail services to identify the recipient of a message

•Support of applications such as automatic call distribution, and enhanced telephony features such as Do Not Disturb and Caller ID

SIP Gateway Support for the Bind Command

•With the bind command, SIP signaling and media paths can advertise the same source IP address on the gateway for certain applications, even if the paths used different addresses to reach the source. This eliminates confusion for firewall applications that, prior to the use of binding, may have taken action on several different source address packets.

SIP Gateway Support of RSVP and TEL URL

•SIP Gateway Support of RSVP and TEL URL enables QoS, ensuring certain bandwidth reservations for specific calls. The bandwidth reservation can be best-effort, in which case the call is completed even if the reservation is not supported by both sides or cannot be established. Or the bandwidth reservation can be required, and the call is not set up if the bandwidth reservation is not performed successfully.

•With the reliable provisional response features, you can ensure that media information is exchanged and resource reservation takes place before connecting a call.

•Forked call responses to Cisco IOS gateways are now supported. Call forking enables the terminating gateway to handle multiple requests and the originating gateway to handle multiple provisional responses for the same call. Call forking is required for the deployment of the find me/follow me type of services.

•Gateways now accept TEL calls sent through the Internet, which provides interoperability with other equipment that uses TEL URL. The TEL URL feature also gives service providers a way to differentiate services based on the type of call, allowing for deployment of specific services.

SIP Intra-Gateway Hairpinning

•Hairpinning enables the same gateway to originate and terminate a call. It works independently of, but enhances, call forking. It also enables call control features to function when it is required that an incoming PSTN call is routed back out to the PSTN on the same Cisco gateway.

SIP INVITE Request with Malformed Via Header

•Incrementing a counter and sending a response, rather than simply discarding the INVITE, if it contains a malformed Via header. The counter provides a useful and immediate indication that an INVITE has been discarded, and the response allows the result to be propagated back to the sender.

SIP T.38 Fax Relay

•Cisco furthers its commitment to open standards and to the success of its customers by supporting standards such as ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks and T-38 Annex-D.

•T.38 Fax Relay over packet networks has become a popular way to bypass tolls associated with sending faxes. SIP T.38 Fax Relay provides standards-based toll bypass for both fax and voice calls. Toll bypass capabilities can result in cost savings to end users of packet telephony networks.

•A Cisco originating gateway (OGW) that has T.38 support automatically enters T.38 mode if it receives a T.38 INVITE, even if it is configured for the Cisco proprietary Fax Relay. This choice of fax protocols provides an extremely reliable fax transfer mechanism.

•Currently, SIP uses the Cisco proprietary Fax Relay solution. However, Cisco Fax Relay is sometimes not an ideal solution for enterprise and service provider customers who have implemented a multivendor network. Because the T.38 Fax Relay protocol is standards-based, Cisco gateways and gatekeepers can operate with third-party T.38-enabled gateways and gatekeepers in a multivendor network where real-time Fax Relay capabilities are required.

•T.38 Fax Relay is already implemented in Cisco gateways that support H.323 and Media Gateway Control Protocol (MGCP). The addition of T.38 for SIP strengthens SIPs position as a low-cost standards-based infrastructure, and increases its viability as the protocol of choice for next-generation IP networks.

SIP User Agent MIB

•The SIP User Agent MIB provides SIP-specific information via SNMP—this information allows customers to have SIP-specific information available to evaluate the performance of gateways in conjunction with their SIP networks.

Restrictions

SIP

Ensure that your access platform has 16 MB Flash memory and 64 MB DRAM memory minimum, and that I/O memory is set to either 8 or 16 MB.

SIP Enhancements

•The SIP Gateway does not support codecs other than those listed in Table 1.

–If on the originating gateway, an appropriate SIP debug trace is presented, indicating the failure to originate the SIP call leg.

–If on the terminating gateway, an appropriate SIP response (4xx) with a warning indicating incompatible media types is sent.

•The SIP Gateway requires each INVITE to include a Session Description Protocol (SDP) header.

•The contents of the Session Description Protocol (SDP) header cannot change between the 180 Ringing message and the 200 OK message.

•SIP requires that all times be sent in Greenwich Mean Time (GMT). The INVITE is sent with GMT. However, the default for routers is to use Coordinated Universal Time (UTC). To configure the router to use GMT, issue the clock timezone command in global configuration mode and specify the GMT time.

•The Enhancements to SIP for VoIP feature supports plain old telephone service (POTS) to POTS hairpinning (which means the call comes in one voice port and is router out another voice port). It also supports POTS to IP call legs and IP to POTS call legs. However, it does not support IP to IP hairpinning. This means the SIP Gateway cannot take an inbound SIP call and reroute it back to another SIP device using the VoIP dial peers.

•SIP requires that all times be sent in Greenwich Mean Time (GMT). The INVITE is sent with GMT. However, the default for routers is to use Coordinated Universal Time (UTC). To configure the router to use GMT, issue the clock timezone command in global configuration mode and specify the GMT.

•VoIP dial peers allow a user to configure the bytes parameter associated with a codec. However, Cisco SIP gateways currently do not present or respond to this parameter. Currently, the a=ptime parameter is not sent or recognized in the SDP body of a SIP message.

•With call transfer, the Requested-By header identifies the party initiating the transfer. The Requested-By header is included in the INVITE request that is sent to the transferred-to party only if a Requested-By header was also included in the Bye request.

•With call transfer, the Also header identifies the transferred-to party. To invoke a transfer, the user portion of the Also header must be defined explicitly or with wildcards as a destination pattern on a VoIP dial peer. The transferred call is routed using the session target parameter on the dial peer instead of the host portion of the Also header. Therefore, the Also header can contain user@host but the host portion is ignored for call routing purposes.

•The grammar for the Also and Requested-By headers is not fully supported. Only the name-addr header is supported. This implies that the crypto-param, which might be present in the Bye request, will not be populated in the ensuing INVITE to the transferred-to party.

•Cisco SIP Gateways do not support the user=np-queried parameter in a Request URI.

•If a Cisco SIP Gateway receives an ISDN Progress message, it generates a 183 Session Progress message. If the gateway receives an ISDN ALERT, it generates a 180 Ringing message.

Call Transfer Capabilities Using the Refer Method

•Although SIP IOS gateways currently support SIP URLs and TEL URLs, the Refer-To header must be in SIP URL format to be valid. The TEL URL format cannot be used, because it does not provide a host portion, and without one, the triggered Invite request cannot be routed.

•Only three overloaded headers in the Refer-To header are accepted: Accept-Contact, Proxy-Authorization, and Replaces. All other headers present in the Refer-To are ignored.

•The Refer-To and Contact headers are required in the Refer request. The absence of either header results in a 4xx class response to the Refer request. Also, the Refer request must contain exactly one Refer-To header. Multiple Refer-To headers result in a 4xx class response.

•The Referred-By header is required in a Refer request. The absence of this header results in a 4xx class response to the Refer request. Also, the Refer request must contain exactly one Referred-By header. Multiple Referred-By headers result in a 4xx class response.

•As with the Bye and Also call transfer methods, the dial peers must be configured for correct functioning of the Refer method.

DTMF Relay for SIP Calls Using Named Telephone Events

•The DTMF Relay for SIP calls Using Named Telephone Events feature is only available on Cisco VoIP gateways using SIP. The DTMF Relay for SIP Calls Using NTE feature does not support hookflash generation for advanced features such as call waiting and conferencing.

SIP Gateway Support of RSVP and TEL URL

•Support for interaction with forking proxies applied only to gateways acting as a user agent client (UAC) is not supported. It does not apply when the gateway acts as a user agent server (UAS). In that case, the proxy forks multiple INVITES with the same call ID to the same gateway but with different request URLs.

•Forking functionality sets up RSVP for each transaction only if the dial peers are configured for QoS. If not, the calls proceed as best-effort. Bandwidth reservation (QoS) is not supported for Session Description Protocol (SDP changes between 183 Session Progress/180 Alerting and 200 OK responses).

•Bandwidth reservation (QoS) is not attempted if the desired QoS level is set to the default of best-effort. The desired QoS for the associated dial peer must be set to controlled-load or guaranteed-delay.

•Distributed Call Signaling (DCS) headers and extensions are not supported.

SIP INVITE Request with Malformed Via Header

•Distributed Call Signaling (DCS) headers and extensions are not supported.

SIP T.38 Fax Relay

•For SIP T.38 Fax Relay only UDP is supported for the transport layer.

•If SIP T.38 Fax Relay is not supported by both gateways, the T.38 negotiation fails and the call reverts back to an audio codec.

•T.38 Fax Relay requires 64 Kbps, the same amount of bandwidth as a voice call with the G.711 codec.

•Calling tones (CNG) are optional, and are not used to initiate a switch to T.38 mode. Instead, called terminal identification tones (CED) or preamble flags are used.

•This feature does not rely on named signaling events (NSE) to signal a switch to T.38 mode. Standard RFC 2543 and RFC 2327 SIP and SDP signaling are used instead.

Note The transport protocols specified in the ITU-T Recommendation for T.38 are Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). However, only UDP is supported for Cisco IOS Release 12.2(2)XB. For further information on T.38 protocol, refer to the ITU-T Recommendations.

Related Features and Technologies

•Cisco Fax Relay

•Cisco IP Phones

•Cisco QoS

•Cisco RSVP

•Cisco SIP Proxy Server

•Cisco TCL/IVR Version 2.0

•Cisco VoIP

Related Documents

The following documents contain information related to Cisco SIP functionality:

•Cisco IOS IP Configuration Guide, Release 12.2

•Cisco IOS IP Command Reference, Volume 1 of 3: Addressing and Services, Release 12.2

•Cisco IOS IP Command Reference, Volume 2 of 3: Routing Protocols, Release 12.2

•Cisco IOS IP Command Reference, Volume 3 of 3: Multicast, Release 12.2

•Cisco IOS Quality of Service Solutions Command Reference, Release 12.2

•Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2

•Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2

•Cisco IP Telephony Network Design Guide

•Configuring Session Initiation Protocol for Voice over IP

•Dial Peer Enhancements

•Service Provider Features for Voice over IP, Release 12.0(3)T

•Session Initiation Protocol Call Flows

•Session Initiation Protocol Gateway Call Flows

•Session Initiation Protocol Gateway Call Flows and Compliance Information

•SIP Call Flows, Release 12.2(4)T

•SIP Diversion Header Implementation for Redirecting Numbe r

•SIP Gateway Support of RSVP and TEL URL, Release 12.2(2)XB

•TCL IVR API Version 2.0 Programmer's Guide

•VoIP Call Admission Control Using RSVP chapter in the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2

•Voice over IP for the Cisco 2600/3600 Series

Supported Platforms

•Cisco 2691

•Cisco 3631

•Cisco 3725

•Cisco 3745

•Cisco 7200 series

•Cisco AS5850

Determining Platform Support Through Cisco Feature Navigator

Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.

Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.

To access Cisco Feature Navigator, you must have an account on Cisco.com. If you have forgotten or lost your account information, send a blank e-mail to cco-locksmith@cisco.com. An automatic check will verify that your e-mail address is registered with Cisco.com. If the check is successful, account details with a new random password will be e-mailed to you. Qualified users can establish an account on Cisco.com by following the directions at http://www.cisco.com/register.

Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:

http://www.cisco.com/go/fn

Supported Standards, MIBs, and RFCs

Standards

•ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks

•ITU-T T.38, Procedures for real-time Group 3 facsimile communication over IP networks, Amendment 1

•ITU-T, T.38, Annex-D

MIBs

•CISCO-SIP-UA-MIB

To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:

http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml

RFCs

•RFC 1890, RTP Profile for Audio and Video Conferences with Minimal Control

•RFC 2327, SIP/SDP Signaling

•RFC 2543, SIP: Session Initiation Protocol

•RFC 2728, A DNS RR for Specifying the Location of Services (DNS SRV)

•RFC 2806, URLs for Telephone Calls

•RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

Prerequisites

General SIP Prerequisites

•Your gateway must have voice functionality that is configurable for either SIP.

•Establish a working IP network.

•Configure VoIP.

•Ensure that your Cisco 2600 series, Cisco 3600 series, or Cisco 7200 series router has 16-MB Flash memory and 64-MB DRAM memory, minimum.

SIP Gateway Support for Bind Command

•Set the bind address prior to using the bind command.

Call Transfer Capabilities Using the Refer Method

•Configure the SIP dial peers for call transfer.

As with the Bye and Also call transfer methods, the dial peers must be configured for correct functioning of the Refer method. See the document Call Transfer Capabilities Using the Refer Method for complete configuration steps.

Configuration Tasks

SIP

See the following sections for configuration tasks for basic SIP functions. Each task in the list is identified as either required or optional.

•Configuring the SIP User Agent (UA) (required)

•Changing the Configuration of the SIP User Agent (UA) (optional)

•Configuring SIP Support for VoIP Dial Peers (optional)

•Configuring a POTS Dial Peer (optional)

•Configuring SIP Call Transfer for a POTS Dial Peer (optional)

•Configuring SIP Call Transfer for a VoIP Dial Peer (optional)

•Configuring Phone Number Translation Rules (required)

•Verifying the SIP Feature Configuration (optional)

For more information on SIP configuration, including call flows, refer to the Session Initiation Protocol Call Flows document.

Call Transfer Capabilities Using the Refer Method

For configuration tasks for this feature, see the document Call Transfer Capabilities Using the Refer Method.

Configurable PSTN Cause Code to SIP Response Mapping

For configuration tasks for this feature, see the document Configurable PSTN Cause Code to SIP Response Mapping.

Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events

For configuration tasks for this feature, see the document Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events.

ISDN Progress Indicator Support for SIP Using 183 Session Progress

There are no configuration tasks for this feature.

RFC 2782 Compliance for DNS SRV

For configuration tasks for this feature, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Diversion Header Implementation for Redirecting Number

For configuration tasks for this feature, see the document SIP Diversion Header Implementation for Redirecting Number.

SIP Gateway Support for Bind Command

For configuration tasks for this feature, see the document SIP Gateway Support for Bind Command.

SIP Gateway Support of RSVP and TEL URL

For configuration tasks for this feature, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Intra-Gateway Hairpinning

There are no configuration tasks for this feature.

SIP INVITE Request with Malformed Via Header

There are no configuration tasks for this feature.

SIP T.38 Fax Relay

For configuration tasks for this feature, see the document SIP T.38 Fax Relay.

SIP User Agent MIB

There are no configuration tasks for this feature.

Configuring the SIP User Agent (UA)

A terminating gateway that is not configured as an SIP user agent cannot receive incoming SIP calls. The transport command opens the SIP listener port (5060) to receive SIP (a SIP user agent is configured to listen by default).

To configure the terminating gateway, enter the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# sip-ua

Enters SIP user-agent mode to configure SIP UA-related commands.

Step 2

Router(config-sip-ua)# transport {udp | tcp}

Configures the SIP user agent (sip-ua) for SIP signaling messages. The default value is udp.

•udp—Configures the SIP user agent to receive SIP messages on UDP port 5060.

•tcp—Configures the SIP user agent to receive SIP messages on TCP port 5060.

Step 3

Router(config-sip-ua)# sip-server ipv4:ip-address

Enters the IP address of the SIP server interface.

Step 4

Router(config-sip-ua)# timers trying number

Sets time to wait for a response.

•number—Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 through 1000. The default is 500.

Step 5

Router(config-sip-ua)# retry invite number

Configures the SIP signaling timers for retry attempts.

•number—Number of INVITE retries: 1 through 10 are valid inputs; default = 6.

Changing the Configuration of the SIP User Agent (UA)

It is not necessary to configure a SIP UA in order to place a call. A SIP UA is configured to listen by default. However, if you want to adjust any of the settings, you can do so using the following commands beginning in global configuration mode:
Command

Purpose

Step 1

Router(config)# sip-ua

Enters SIP user agent (sip-ua) mode to configure SIP UA-related commands.

Step 2

Router(config-sip-ua)# transport {udp | tcp}

Configures the SIP user agent (sip-ua) for SIP signaling messages. The default value is udp.

•udp—Configures the SIP user agent to receive SIP messages on UDP port 5060.

•tcp—Configures the SIP user agent to receive SIP messages on TCP port 5060.

Step 3

Router(config-sip-ua)# sip-server {dns: host-name | ipv4:ip-address [port-number]}

Enters the host name or IP address of the SIP server interface.

•dns:—Sets the global SIP server interface to a DNS.

•host-name—A valid DNS host name takes the following format: gateway.company.com.

•ipv4:ip-address—Sets the global SIP server interface to an IP address. A valid IP address takes the following format: xxx.xxx.xxx.xxx.

•port-number—(Optional) Specifies the port number for the SIP server.

Step 4

Router(config-sip-ua)# timers trying number

Sets time to wait for a response.

•number—Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 through 1000. The default is 500.

Step 5

Router(config-sip-ua)# timers expires number

Limits the time duration (in milliseconds) of a search for an INVITE.

•number—Specifies the time (in milliseconds) for which an INVITE request is valid. Possible values are 60000 through 300000. The default is 180000.
Step 6
Router(config-sip-ua)# retry invite 
number
Configures the SIP signaling timers for retry attempts.

•number—Specifies the number of INVITE retries. Valid values are 1 through 10. The default is 6.
Step 7
Router(config-sip-ua)# max-forwards 
number
Limits the number of proxy or redirect servers that can forward a request.

•number—Number of hops. Valid values are 1 through 15. The default is 6.
Configuring SIP Support for VoIP Dial Peers

To configure SIP support for a VoIP dial peer, enter the following commands beginning in global configuration mode:
Command

Purpose

Step 1

Router(config)# dial-peer voice tag voip

Enters dial-peer configuration mode to configure a VoIP dial peer.

•tag—Digits that define a particular dial peer. Valid entries are from 1 to 2,147,483,647.

Step 2

Router(config-dial-peer)# destination-pattern [+]string[T]

Defines the telephone number associated with this VoIP dial peer.

•+—(Optional) Character indicating an E.164 standard number.

•string—Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9, the letters A through D, and the following special characters:

–The asterisk (*) and pound sign (#) that appear on standard touch-tone dial pads. On the Cisco 3600 series routers only, these characters cannot be used as leading characters in a string (for example, *650).

–Comma (,), which inserts a pause between digits.

–Period (.), which matches any entered digit (this character is used as a wildcard). On the Cisco 3600 series routers, the period cannot be used as a leading character in a string (for example, .650).

–Percent sign (%), which indicates that the previous digit/pattern occurred zero or multiple times, similar to the wildcard usage in the regular expression.

–Plus sign (+), which matches a sequence of one or more matches of the character/pattern.

Note The plus sign used as part of the digit string is different from the plus sign that can be used in front of the digit string to indicate that the string is an E.164 standard number.
–Circumflex (^), which indicates a match to the beginning of the string.

–Dollar sign ($), which matches the null string at the end of the input string.

–Backslash symbol (\), which is followed by a single character matching that character or used with a single character with no other significance (matching that character).

–Question mark (?), which indicates that the previous digit occurred zero or one time.

–Brackets ( [ ] ), which indicate a range. A range is a sequence of characters enclosed in the brackets; only numeric characters from 0 to 9 are allowed in the range. This is similar to a regular expression rule.

–Parentheses "( )", which indicate a pattern and is the same as the regular expression rule.

•T—(Optional) Control character indicating that the destination-pattern value is a variable length dial string.
Step 3

Router(config-dial-peer)# session transport {udp | tcp}

Enters the session transport type for the SIP user agent.

•udp—Configures the SIP user agent to receive SIP messages on UDP port 5060.

•tcp—Configures the SIP user agent to receive SIP messages on TCP port 5060.

Step 4

Router(config-dial-peer)# session protocol sipv2

Enters the session protocol type as IETF Session Inititation Protocol.

Step 5

Router(config-dial-peer)# session target sip-server

Specifies the dial peer session target to use the global SIP server.
Configuring a POTS Dial Peer

To configure a POTS dial peer, enter the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# dial-peer voice tag voip

Enters dial-peer configuration mode to configure a VoIP dial peer.

Step 2

Router(config-dial-peer)# destination-pattern [+]string[T]

Defines the telephone number associated with this POTS dial peer.

Step 3

Router(config-dial-peer)# port slot-number/subunit-number/port

Associates this POTS dial peer with a specific voice port.

Step 4

Router(config-dial-peer)# session transport {udp | tcp}

Enters the session transport type for the SIP user agent.

Step 5

Router(config-dial-peer)# session protocol sipv2

Enters the session protocol type.

Step 6

Router(config-dial-peer)# session target sip-server

Specifies the dial peer session target to use the global SIP server.

Configuring SIP Call Transfer for a POTS Dial Peer

To configure SIP call transfer for a POTS dial peer, enter the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# dial-peer voice tag pots

Enters dial-peer configuration mode to configure a POTS dial peer.

Step 2

Router(config-dial-peer)# application session

Specifies that the standard session application will be invoked for this dial peer.

Step 3

Router(config-dial-peer)# destination-pattern [+]string[T]

Specifies the telephone number associated with the dial peer.

Step 4

Router(config-dial-peer)# port slot/port

Specifies the voice slot number and port through which incoming VoIP calls are received.

Configuring SIP Call Transfer for a VoIP Dial Peer

To configure SIP call transfer for a VoIP dial peer, enter the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# dial-peer voice tag voip

Enters dial-peer configuration mode to configure a VoIP dial peer.

Step 2

Router(config-dial-peer)# application session

Specifies that the standard session application will be invoked for this dial peer.

Step 3

Router(config-dial-peer)# destination-pattern [+]string[T]

Specifies the telephone number associated with the dial peer.

Step 4

Router(config-dial-peer)# session target ipv4:ip-address

Specifies the IP address of the destination gateway for outbound dial peers.

Configuring Phone Number Translation Rules

By default, the SIP gateway tags called numbers that have 11 or more digits as "international" when sending SETUP messages to the PSTN switch. In some cases, such as situations where the user must dial 9 to access an outside line, this assumption may not be correct.

To accommodate such situations, you can define translation rules on the outbound POTS dial peer to convert the "type of number" to the correct value. Translation rules manipulate the called number digits and the "type of number" value associated with the called digits.

To define translation rules on a POTS dial peer, enter the following commands beginning in global configuration mode:

Command

Purpose

Step 1

Router(config)# translation-rule name-tag

Defines a translation-rule tag number and enters translation-rule configuration mode. All subsequent commands that you enter in this mode before you exit will apply to this translation-rule tag.

•name-tag—The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. The range is 1 through 2,147,483,647.

Step 2

Router(config-translate)# rule name-tag input-matched-pattern substituted-pattern [match-type substituted-type]

Specifies translation rules. This command can be entered multiple times and is applied to the translation-rule defined in Step 1.

•name-tag—The tag number by which the rule set will be referenced. This is an arbitrarily chosen number. Range is from 1 through 2,147,483,647.

•input-matched-pattern— The input string of digits for which pattern matching is performed.

•substituted-pattern—The replacement digit string that results after pattern matching is performed. Regular expressions are used to carry out this process.

•match-type—(Optional) The choices for this field are, abbreviated, any, international, national, reserved, subscriber, and unknown, as defined by the International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Q.931 specification. If you enter the match-type value, then you must also enter the substituted-type value.

•substituted-type—(Optional) The choices for this field are abbreviated, international, national, reserved, subscriber, and unknown, as defined by the ITU Q.931 specification.

Step 3

Router(config-translate)# exit

Exits from translate configuration mode.

Step 4

Router(config)# dial-peer voice tag pots

Enter the dial-peer mode to configure a POTS dial peer.

Step 5

Router(config-dial-peer)# translate-outgoing called name-tag

Specifies the translation tag for an outbound called number.

•name-tag—Translation rule tag. Valid values are 1 to 2,147,483,647.

Step 6

Router(config-dial-peer)# port slot-number/port

Specifies the voice port.

For more information about the commands used to configure translation rules, see the Dial Peer Enhancements documentation on Cisco.com.

Verifying the SIP Feature Configuration

Enter the show running configuration command to verify your configuration.

Configuration Examples

This section provides the following configuration examples:

•Basic SIP Configuration Example

•Configuring SIP with Multiple Codecs Example

•Configuring Phone Number Translation Rules Examples

•Call Transfer Configuration Examples

Call Transfer Capabilities Using the Refer Method

For configuration examples for this feature, see the document Call Transfer Capabilities Using the Refer Method.

Configurable PSTN Cause Code to SIP Response Mapping

For configuration examples for this feature, see the document Configurable PSTN Cause Code to SIP Response Mapping.

Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events

For configuration examples for this feature, see the document Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events.

ISDN Progress Indicator Support for SIP Using 183 Session Progress

There are no configuration examples for this feature.

RFC 2782 Compliance for DNS SRV

For configuration examples for this feature, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Diversion Header Implementation for Redirecting Number

For configuration examples for this feature, see the document SIP Diversion Header Implementation for Redirecting Number.

SIP Gateway Support for Bind Command

For configuration examples for this feature, see the document SIP Gateway Support for Bind Command.

SIP Gateway Support of RSVP and TEL URL

For configuration examples for this feature, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Intra-Gateway Hairpinning

There are no configuration examples for this feature.

SIP INVITE Request with Malformed Via Header

There are no configuration examples for this feature.

SIP T.38 Fax Relay

For configuration examples for this feature, see the document SIP T.38 Fax Relay.

SIP User Agent MIB

There are no configuration examples for this feature.

Basic SIP Configuration Example

The following shows an example of the output that appears when you enter the show running configuration command.
Router1# show running configuration
Building configuration...
Current configuration:
!
version 12.2
service timestamps debug datetime
service timestamps log uptime
no service password-encryption
!
hostname router1
!
enable secret 5 $1$dlEK$ziROgcQm08RwI/d0VSfal1
enable password password1
!
dspint DSPfarm1/0
!
ip subnet-zero
ip tcp path-mtu-discovery
ip name-server 172.18.192.48
!
isdn voice-call-failure 0
!
!
controller T1 1/0
 framing esf
 clock source line primary
 linecode b8zs
!
controller T1 1/1
!
!
voice-port 2/0/0
!
voice-port 2/0/1
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g723r63
 codec preference 3 g723r53
!!
dial-peer voice 100 pots
 destination-pattern 3660110
 port 2/0/0
!
dial-peer voice 200 pots
 application session
 destination-pattern 3660120
 port 2/0/1
!
dial-peer voice 101 voip
 destination-pattern 3660210
 session protocol sipv2
 session target ipv4:166.34.244.73
 codec g711ulaw
!
dial-peer voice 201 voip
 application sesion
 destination-pattern 3660220
 session protocol sipv2
 session target dns:3660-2.sip.com
 codec g711ulaw
!
dial-peer voice 999 voip
 destination-pattern 5551111
 session protocol sipv2
 session target ipv4:161.44.53.89
 session transport tcp
!
dial-peer voice 300 pots
 destination-pattern 2101100
!
dial-peer voice 350 voip
 destination-pattern 3100607
 session protocol sipv2
 session target ipv4:172.18.192.197
 codec g711ulaw
!
dial-peer voice 301 voip
 application session
 destination-pattern 1234
 session protocol sipv2
 session target ipv4:172.18.192.193
 codec g711ulaw
!
dial-peer voice 333 voip
 application session
 destination-pattern 1235
 session protocol sipv2
 session target ipv4:172.18.192.199
 codec g711ulaw
!
dial-peer voice 888 voip
 destination-pattern 888
 session protocol sipv2
 session target ipv4:161.44.53.89
 session transport tcp
 codec g711ulaw
!
dial-peer voice 260011 voip
 destination-pattern 260011
 session protocol sipv2
 session target ipv4:172.18.192.164
 codec g711ulaw
!
dial-peer voice 444 voip
 destination-pattern 2339000
 session protocol sipv2
 session target ipv4:172.18.192.205
 codec g711ulaw
!
dial-peer voice 111 voip
 destination-pattern 111
 session protocol sipv2
 session target sip-server
 codec g711ulaw
!
dial-peer voice 7777777 voip
 destination-pattern 19197777777
 session protocol sipv2
 session target ipv4:172.18.192.38
 codec g711ulaw
!
!
sip-ua 
 max-forwards 0
 retry invite 5
 retry response 0
 retry bye 0
 retry cancel 0
 retry prack 0
 retry comet 0
 retry rel1xx 0
 retry notify 0
 timers trying 501
 timers expires 0
 timers connect 0
 timers disconnect 0
 timers prack 0
 timers comet 0
 timers rel1xx 0
 timers notify 0
 sip-server ipv4:172.16.0.0
 no transport tcp!
!
interface FastEthernet0/0
 ip address 172.18.192.194 255.255.255.0
 load-interval 30
 speed auto
 half-duplex
!
interface FastEthernet0/1
 ip address 166.34.245.230 255.255.255.224
 load-interval 30
 speed auto
 half-duplex
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.18.192.1
ip route 166.34.0.0 255.255.0.0 166.34.245.225
no ip http server
!
access-list 101 permit ip host 10.0.2.30 host 10.0.2.31
access-list 101 deny   udp any eq rip any
access-list 101 deny   udp any any eq rip
access-list 101 deny   udp any eq isakmp any
access-list 101 deny   udp any any eq isakmp
access-list 101 permit ip any any
snmp-server engineID local 000000090200003094202740
snmp-server community public RW
!         
line con 0
 exec-timeout 0 0
 transport input none
line aux 0
line vty 0 4
 password xxx
 login
!
end
Configuring SIP with Multiple Codecs Example

The following shows an example of the output that appears when you enter the show running configuration command. Inapplicable modules are omitted.
Router# show running-configuration
version 12.2
. 
. 
.
hostname UA-4
. 
. 
.
controller T1 0
 framing esf
 clock source line primary
 linecode b8zs
 ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis
. 
. 
.
controller T1 1
 framing esf
 clock source line secondary 1
 linecode b8zs
 ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis
. 
. 
.
voice-port 0:0
. 
. 
.
voice-port 1:0
. 
. 
.
voice class codec 100
 codec preference 1 g726r16
 codec preference 2 g729r8
 codec preference 3 g711alaw
 codec preference 4 g711ulaw
. 
. 
.
dial-peer voice 500 pots
 destination-pattern 92055500..
 port 0:0
 prefix 92055500
. 
. 
.
dial-peer voice 600 voip
 incoming called-number 92055500..
 session protocol sipv2
 voice-class codec 100
 no vad
. 
. 
.
dial-peer voice 501 pots
 destination-pattern 94055500..
 port 1:0
 prefix 94055500
. 
. 
.
dial-peer voice 601 voip
 incoming called-number 94055500..
 session protocol sipv2
 voice-class codec 100
 no vad
. 
. 
.
interface Ethernet0
 ip address 172.16.1.1 255.255.255.1
 no ip directed-broadcast
 load-interval 30
. 
. 
.
interface FastEthernet0
 ip address 172.16.1.2 255.255.255.2
 no ip directed-broadcast
 load-interval 30
 duplex auto
 speed auto
Configuring Phone Number Translation Rules Examples

The following example illustrates a translation rule for dialing national numbers in the situation where the user must dial 9 to access an outside line. In the rule command in this example:

•91% is the input search pattern. The percent sign (%) is a wild card.

•The second 1 is the substituted pattern.

•The match type of number is international.

•The substituted type of number is national.

The result of this command is that any outgoing call that is destined for a number that starts with 91 and that is considered by the gateway to be an international number will be sent to the PSTN as a national number with a prefix of 1.
translation-rule 10
Rule 1 91% 1 international national
!
!
!
dial-peer voice 10 pots
destination-pattern 91..........
translate-outgoing called 10
port 1:D
!
The following example illustrates a translation rule for dialing national numbers in the situation where the user does not need to dial 9 to access an outside line.
translation-rule 10
Rule 1 1% 1 international national
!
!
!
dial-peer voice 10 pots
destination-pattern 1..........
translate-outgoing called 10
port 1:D
prefix 1
!
The following example illustrates a translation rule for dialing international numbers in the situation where the user must dial 9 to access an outside line.
translation-rule 20
Rule 1 9011% 011 unknown international
!
!
!
dial-peer voice 10 pots
destination-pattern 9011T
translate-outgoing called 20
port 1:D
!
The following example illustrates a translation rule for dialing international numbers in the situation where the user does not need to dial 9 to access an outside line.
translation-rule 20
Rule 1 011% 011 unknown international
!
!
!
dial-peer voice 10 pots
destination-pattern 011T
translate-outgoing called 20
port 1:D
prefix 011
!
Call Transfer Configuration Examples

The following example illustrates how to configure call transfer. In Figure 1, User A and User C are in an established call. User C then transfers the call to User B. This results in call establishment between User A and User B. User C is then disconnected with User A, regardless of whether the transfer fails or succeeds.

When a call originates or terminates on a gateway, either the calling party number, the called party number, or the port is used (depending on the scenario) to match a dial peer in order to determine the basic call characteristics. One of the characteristics to determine is which application to use for the call. For the call transfer to succeed, the matching dial peer must have application set to "session" on the gateway that is controlling the transfer. (This is the gateway that receives the Bye with an Also header).

There are two scenarios for dial-peer matching based on whether the call is coming from a POTS interface or from the IP network:

•For calls coming from a POTS interface, the port will be used to match a POTS dial peer with the port the call came in from. This dial peer should have "application session."

•For calls coming from the IP network, a series of criteria is used (in the order listed below) to match dial peers. If the first criteria does not result in a match, the second criteria is used. If the second criteria does not result in a match, the third criteria is used. If a match does not occur, the default application, which does not support call transfer, is used.

a. The called number matches the "incoming called-number" on a VoIP dial peer.

b. The calling number matches the "answer-address" on a VoIP dial peer.

c. The calling number matches the "destination-pattern" on a VoIP dial peer.

Note For calls coming from the IP network, it is possible for the calling number to be blocked based on privacy restrictions. In such cases, the "incoming called-number" is used for call transfers.

Figure 1 Call Transfer Example

In this example, Gateway 1 handles the transfer (recipient of the Bye with the Also header). User C invokes the transfer service (originator of the Bye with the Also header). There are two scenarios in which a dial peer match must have application set to "session" for the transfer to succeed:

•Incoming call from the PSTN—User A originates a call to User C. From the prospective of Gateway 1, this would be an incoming call from the POTS interface so Gateway 1 looks for a POTS dial peer matching the port on which the call came in. Gateway 1 must have a POTS dial peer for User A with application set to "session" if transfer is later invoked by User C.

•Incoming call from IP network—User C calls User A. From the prospective of Gateway 1 this is an incoming call from the IP network. Gateway 1 uses the criteria previously discussed for a VoIP dial peer (match on incoming called-number, answer-address, or destination pattern). Gateway 1 must have one of the following:

–A VoIP dial peer with an incoming called-number of User A

–A VoIP dial peer with answer-address of User C

–A VoIP dial peer with destination-pattern of User C.

The matching dial peer must have application set to "session" if transfer is later invoked by User C.

Note To handle all call transfer situations, you should configure both POTS and VoIP dial peers.

The following example shows how to apply the "session" application to a dial peer:
Router(config)# dial-peer voice 10 pots
Router(config-dial-peer)# application session
The following example shows how to configure the E.164 telephone number 555-7922 for a dial peer:
Router(config)# dial-peer voice 10 pots
Router(config-dial-peer)# destination-pattern +5557922
The following example configures the number (310) 555-9261 as the incoming called number for VoIP dial peer 10:
Router(config)# dial-peer voice 10 pots
Router(config-dial-peer)# incoming called-number 3105559261
The following example configures the E.164 telephone number 555-9626 as the dial peer of an incoming call:
Router(config)# dial-peer voice 10 pots
Router(config-dial-peer)# answer-address +5559626
Command Reference

This section documents modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.2 command reference publications and in the following documents:

Call Transfer Capabilities Using the Refer Method

For command reference pages for this feature, see the document Call Transfer Capabilities Using the Refer Method.

Configurable PSTN Cause Code to SIP Response Mapping

For command reference pages for this feature, see the document Configurable PSTN Cause Code to SIP Response Mapping.

Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events

For command reference pages for this feature, see the document Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events.

ISDN Progress Indicator Support for SIP Using 183 Session Progress

There are no commands associated with this feature.

RFC 2782 Compliance for DNS SRV

For command reference pages for this feature, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Diversion Header Implementation for Redirecting Number

For command reference pages for this feature, see the document SIP Diversion Header Implementation for Redirecting Number.

SIP Gateway Support for Bind Command

For command reference pages for this feature, see the document SIP Gateway Support for Bind Command.

SIP Gateway Support of RSVP and TEL URL

For command reference pages for this feature, see the document SIP Gateway Support of RSVP and TEL URL.

SIP Intra-Gateway Hairpinning

There are no commands associated with this feature.

SIP INVITE Request with Malformed Via Header

There are no commands associated with this feature.

SIP T.38 Fax Relay

For command reference pages for this feature, see the document SIP T.38 Fax Relay.

SIP User Agent MIB

There are no commands associated with this feature.

•debug ccsip all

•debug ccsip calls

•debug ccsip error

•debug ccsip events

•debug ccsip messages

•debug ccsip states

•default

•gw-accounting

•gw-accounting

•max-redirects

•retry invite

•retry invite

•session protocol

•session transport

•show sip-ua

•sip-server

•sip-ua

•timers

•transport

aaa username

To determine the information to populate the username attribute for AAA billing records, use the aaa username command in SIP user agent configuration mode. To achieve default capabilities, use the no form of this command.

aaa username {calling-number | proxy-auth}
no aaa username
Syntax Description

calling-number

Uses the FROM: header in the SIP INVITE (default value). This keyword is used in most implementations.

proxy-auth

Parses the Proxy-Authorization header. Decodes the Microsoft Passport user ID (PUID) and password, and then populates the PUID into the username attribute and a "." into the password attribute.

The username attribute is used for billing and the "." is used for the password, because the user has already been authenticated prior to this point.

Defaults

calling-number

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.2(2)X

This command was introduced.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

Parsing of the Proxy-Authorization header, decoding of the PUID and password, and populating of the username attribute with the PUID must be enabled through this command. If this command is not issued, the Proxy-Authorization header is ignored.

The keyword proxy-auth is a nonstandard implementation, and SIP gateways do not normally receive or process the proxy-auth header.

Examples

The following example shows the processing of the SIP username from the Proxy-Authorization header being enabled:
Router(config)# sip-ua
Router(config-sip-ua)# aaa username proxy-auth
Related Commands

Command

Description

show call active voice

Shows active call information for voice calls or fax transmissions in progress.

show call history voice

Displays the voice call history table.

debug ccsip all

To enable all SIP-related debugging, enter the debug ccsip all command in EXEC mode. To disable all debugging output, use the no form of this command.

debug ccsip all
no debug ccsip all
Syntax Description

This command has no arguments or keywords.

Defaults

SIP debugging is not enabled

Command Modes

EXEC

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

The output of this command was changed.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

The debug ccsip all command enables the following debug SIP commands:

Command

Description

debug ccsip calls

Shows all SIP Service Provider Interface (SPI) call tracing.

debug ccsip error

Shows SIP Service Provider Interface (SPI) errors.

debug ccsip events

Shows all SIP Service Provider Interface (SPI) events tracing.

debug ccsip messages

Shows all SIP Service Provider Interface (SPI) message tracing.

debug ccsip states

Shows all SIP Service Provider Interface (SPI) state tracing.

Examples

The following example displays debug output from one side of the call:
Router# debug ccsip all
All SIP call tracing enabled
Router1#
*Mar  6 14:10:42: 0x624CFEF8 : State change from (STATE_NONE, SUBSTATE_NONE)  to 
(STATE_IDLE, SUBSTATE_NONE)
*Mar  6 14:10:42:  Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  act_idle_call_setup
*Mar  6 14:10:42:  act_idle_call_setup:Not using Voice Class Codec
*Mar  6 14:10:42: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160 
*Mar  6 14:10:42:  Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
*Mar  6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_NONE)  to 
(STATE_IDLE, SUBSTATE_CONNECTING)
*Mar  6 14:10:42: REQUEST CONNECTION TO IP:166.34.245.231 PORT:5060
*Mar  6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_CONNECTING)  to 
(STATE_IDLE, SUBSTATE_CONNECTING)
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  act_idle_connection_created
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  act_idle_connection_created: Connid(1) created to 
166.34.245.231:5060, local_port 54113
*Mar  6 14:10:42: sipSPIAddLocalContact
*Mar  6 14:10:42:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_CONNECTING)  to 
(STATE_SENT_INVITE, SUBSTATE_NONE)
*Mar  6 14:10:42: Sent: 
INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Cisco-Guid: 2881152943-2184249548-0-483039712
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731427042
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar  6 14:10:42: Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Length: 0
*Mar  6 14:10:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:5060
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  act_sentinvite_new_message
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:10:42:  Roundtrip delay 4 milliseconds for method INVITE
*Mar  6 14:10:42: 0x624CFEF8 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to 
(STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
*Mar  6 14:10:42: Received: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar  6 14:10:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:5060
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  act_recdproc_new_message
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  sipSPICheckResponse : Updating session description
*Mar  6 14:10:42: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:10:42:  Roundtrip delay 8 milliseconds for method INVITE
*Mar  6 14:10:42: HandleSIP1xxRinging: SDP MediaTypes negotiation successful!
Negotiated Codec      : g711ulaw , bytes :160
Inband Alerting       : 0 
*Mar  6 14:10:42: 0x624CFEF8 : State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_PROCEEDING)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING)
*Mar  6 14:10:46: Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar  6 14:10:46: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:5060
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  act_recdproc_new_message
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  sipSPICheckResponse : Updating session description
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:10:46:  Roundtrip delay 3536 milliseconds for method INVITE
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  act_recdproc_new_message: SDP MediaTypes negotiation 
successful!
Negotiated Codec      : g711ulaw , bytes :160
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  sipSPIReconnectConnection
*Mar  6 14:10:46:  Queued event from SIP SPI : SIPSPI_EV_RECONNECT_CONNECTION
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  recv_200_OK_for_invite
*Mar  6 14:10:46:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  6 14:10:46: 0x624CFEF8 : State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_ALERTING)  to (STATE_ACTIVE, SUBSTATE_NONE)
*Mar  6 14:10:46: The Call Setup Information is :
        Call Control Block (CCB) : 0x624CFEF8
         State of The Call        : STATE_ACTIVE
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.230
         Source IP Port    (Media): 20208
         Destn  IP Address (Media): 166.34.245.231
         Destn  IP Port    (Media): 20038
         Destn SIP Addr (Control) : 166.34.245.231
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.231
*Mar  6 14:10:46: HandleUdpReconnection: Udp socket connected for fd: 1 with 
166.34.245.231:5060
*Mar  6 14:10:46: Sent: 
ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Max-Forwards: 6
Content-Type: application/sdp
Content-Length: 137
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  ccsip_caps_ind
*Mar  6 14:10:46: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Mar  6 14:10:46: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE
*Mar  6 14:10:46: CCSIP-SPI-CONTROL:  ccsip_caps_ack
*Mar  6 14:10:50: Received: 
BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Mon, 08 Mar 1993 22:36:44 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 731612207
CSeq: 101 BYE
Content-Length: 0
*Mar  6 14:10:50: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:54835
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  act_active_new_message
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  sact_active_new_message_request
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  6 14:10:50:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  sipSPIInitiateCallDisconnect : Initiate call 
disconnect(16) for outgoing call
*Mar  6 14:10:50: 0x624CFEF8 : State change from (STATE_ACTIVE, SUBSTATE_NONE)  to 
(STATE_DISCONNECTING, SUBSTATE_NONE)
*Mar  6 14:10:50: Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Sat, 06 Mar 1993 19:10:50 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Timestamp: 731612207
Content-Length: 0
CSeq: 101 BYE
*Mar  6 14:10:50:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  act_disconnecting_disconnect
*Mar  6 14:10:50: CCSIP-SPI-CONTROL:  sipSPICallCleanup
*Mar  6 14:10:50:  Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
*Mar  6 14:10:50: CLOSE CONNECTION TO CONNID:1
*Mar  6 14:10:50: sipSPIIcpifUpdate :CallState: 4 Playout: 1755 DiscTime:48305031 ConnTime 
48304651
*Mar  6 14:10:50: 0x624CFEF8 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to 
(STATE_DEAD, SUBSTATE_NONE)
*Mar  6 14:10:50: The Call Setup Information is :
        Call Control Block (CCB) : 0x624CFEF8
         State of The Call        : STATE_DEAD
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.230
         Source IP Port    (Media): 20208
         Destn  IP Address (Media): 166.34.245.231
         Destn  IP Port    (Media): 20038
         Destn SIP Addr (Control) : 166.34.245.231
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.231
*Mar  6 14:10:50: 
        Disconnect Cause (CC)    : 16
        Disconnect Cause (SIP)   : 200
*Mar  6 14:10:50: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote 
port: 5060
The following example displays debut output from the other side of the call:
Router# debug ccsip all
All SIP call tracing enabled
3660-2#
*Mar  8 17:36:40: Received: 
INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Cisco-Guid: 2881152943-2184249548-0-483039712
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731427042
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar  8 17:36:40: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.230:54113
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  sipSPISipIncomingCall
*Mar  8 17:36:40: 0x624D8CCC : State change from (STATE_NONE, SUBSTATE_NONE)  to 
(STATE_IDLE, SUBSTATE_NONE)
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  act_idle_new_message
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  sact_idle_new_message_invite
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  8 17:36:40:  sact_idle_new_message_invite:Not Using Voice Class Codec
*Mar  8 17:36:40: sact_idle_new_message_invite: Preferred codec[0] type: g711ulaw Bytes 
:160
*Mar  8 17:36:40: sact_idle_new_message_invite: Media Negotiation successful for an
incoming call
*Mar  8 17:36:40: sact_idle_new_message_invite: Negotiated Codec      : g711ulaw, bytes 
:160
Preferred Codec       : g711ulaw, bytes :160
*Mar  8 17:36:40:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:36:40: Num of Contact Locations 1 3660110 166.34.245.230 5060
*Mar  8 17:36:40: 0x624D8CCC : State change from (STATE_IDLE, SUBSTATE_NONE)  to 
(STATE_RECD_INVITE, SUBSTATE_RECD_INVITE_CALL_SETUP)
*Mar  8 17:36:40: Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Length: 0
*Mar  8 17:36:40:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  act_recdinvite_proceeding
*Mar  8 17:36:40:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_ALERTING
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  ccsip_caps_ind
*Mar  8 17:36:40: ccsip_caps_ind: codec(negotiated) = 5(Bytes 160)
*Mar  8 17:36:40: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Mar  8 17:36:40: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  ccsip_caps_ack
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  act_recdinvite_alerting
*Mar  8 17:36:40:  180 Ringing with SDP - not likely
*Mar  8 17:36:40:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:36:40: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:36:40: 0x624D8CCC : State change from (STATE_RECD_INVITE, 
SUBSTATE_RECD_INVITE_CALL_SETUP)  to (STATE_SENT_ALERTING, SUBSTATE_NONE)
*Mar  8 17:36:40: Sent: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar  8 17:36:44:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_CONNECT
*Mar  8 17:36:44: CCSIP-SPI-CONTROL:  act_sentalert_connect
*Mar  8 17:36:44: sipSPIAddLocalContact
*Mar  8 17:36:44:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:36:44: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:36:44: 0x624D8CCC : State change from (STATE_SENT_ALERTING, SUBSTATE_NONE)  to 
(STATE_SENT_SUCCESS, SUBSTATE_NONE)
*Mar  8 17:36:44: Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar  8 17:36:44: Received: 
ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Max-Forwards: 6
Content-Type: application/sdp
Content-Length: 137
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar  8 17:36:44: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.230:54113
*Mar  8 17:36:44: CCSIP-SPI-CONTROL:  act_sentsucc_new_message
*Mar  8 17:36:44: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  8 17:36:44: 0x624D8CCC : State change from (STATE_SENT_SUCCESS, SUBSTATE_NONE)  to 
(STATE_ACTIVE, SUBSTATE_NONE)
*Mar  8 17:36:44: The Call Setup Information is :
        Call Control Block (CCB) : 0x624D8CCC
         State of The Call        : STATE_ACTIVE
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.231
         Source IP Port    (Media): 20038
         Destn  IP Address (Media): 166.34.245.230
         Destn  IP Port    (Media): 20208
         Destn SIP Addr (Control) : 166.34.245.230
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.230
*Mar  8 17:36:47:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  act_active_disconnect
*Mar  8 17:36:47:  Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
*Mar  8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_NONE)  to 
(STATE_ACTIVE, SUBSTATE_CONNECTING)
*Mar  8 17:36:47: REQUEST CONNECTION TO IP:166.34.245.230 PORT:5060
*Mar  8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING)  to 
(STATE_ACTIVE, SUBSTATE_CONNECTING)
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  act_active_connection_created
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sipSPICheckSocketConnection
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sipSPICheckSocketConnection: Connid(1) created to 
166.34.245.230:5060, local_port 54835
*Mar  8 17:36:47:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING)  to 
(STATE_DISCONNECTING, SUBSTATE_NONE)
*Mar  8 17:36:47: Sent: 
BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Mon, 08 Mar 1993 22:36:44 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 731612207
CSeq: 101 BYE
Content-Length: 0
*Mar  8 17:36:47: Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Sat, 06 Mar 1993 19:10:50 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Timestamp: 731612207
Content-Length: 0
CSeq: 101 BYE
*Mar  8 17:36:47: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.230:54113
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  act_disconnecting_new_message
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sact_disconnecting_new_message_response
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:36:47:  Roundtrip delay 4 milliseconds for method BYE
*Mar  8 17:36:47: CCSIP-SPI-CONTROL:  sipSPICallCleanup
*Mar  8 17:36:47:  Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
*Mar  8 17:36:47: CLOSE CONNECTION TO CONNID:1
*Mar  8 17:36:47: sipSPIIcpifUpdate :CallState: 4 Playout: 1265 DiscTime:66820800 ConnTime 
66820420
*Mar  8 17:36:47: 0x624D8CCC : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to 
(STATE_DEAD, SUBSTATE_NONE)
*Mar  8 17:36:47: The Call Setup Information is :
        Call Control Block (CCB) : 0x624D8CCC
         State of The Call        : STATE_DEAD
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.231
         Source IP Port    (Media): 20038
         Destn  IP Address (Media): 166.34.245.230
         Destn  IP Port    (Media): 20208
         Destn SIP Addr (Control) : 166.34.245.230
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.230
*Mar  8 17:36:47: 
        Disconnect Cause (CC)    : 16
        Disconnect Cause (SIP)   : 200
*Mar  8 17:36:47: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote 
port: 5060
Related Commands

Command

Description

debug ccsip calls

Displays all SIP SPI call tracing and traces the SIP call details as they are updated in the SIP call control block.

debug ccsip error

Displays SIP SPI errors. This command traces all error messages generated from errors encountered by the SIP subsystem.

debug ccsip events

Displays all SIP SPI events tracing and traces the events posted to SIP SPI from all interfaces.

debug ccsip messages

Displays all SIP SPI message tracing and traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.

debug ccsip states

Displays all SIP SPI state tracing and traces the state machine changes of SIP SPI and displays the state transitions.

debug ccsip calls

To display all SIP SPI call tracing and to trace the SIP call details as they are updated in the SIP call control block, enter the debug ccsip calls command in EXEC mode.

debug ccsip calls
Syntax Description

This command has no arguments or keywords.

Defaults

SIP SPI call tracing is not displayed

Command Modes

EXEC

Command History

Release

Release

12.1(1)T

This command was introduced.

12.1(3)T

The output of the command was changed.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

This command traces the SIP call details as updated in the SIP call control block.

Examples

The following example displays debug output from one side of the call:
Router1# debug ccsip calls
SIP Call statistics tracing is enabled
Router1#
*Mar  6 14:12:33: The Call Setup Information is :
        Call Control Block (CCB) : 0x624D078C
         State of The Call        : STATE_ACTIVE
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.230
         Source IP Port    (Media): 20644
         Destn  IP Address (Media): 166.34.245.231
         Destn  IP Port    (Media): 20500
         Destn SIP Addr (Control) : 166.34.245.231
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.231
*Mar  6 14:12:40: The Call Setup Information is :
        Call Control Block (CCB) : 0x624D078C
         State of The Call        : STATE_DEAD
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.230
         Source IP Port    (Media): 20644
         Destn  IP Address (Media): 166.34.245.231
         Destn  IP Port    (Media): 20500
         Destn SIP Addr (Control) : 166.34.245.231
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.231
*Mar  6 14:12:40: 
        Disconnect Cause (CC)    : 16
        Disconnect Cause (SIP)   : 200
The following example displays debug output from the other side of the call:
Router2# debug ccsip calls
SIP Call statistics tracing is enabled
Router2#
*Mar  8 17:38:31: The Call Setup Information is :
        Call Control Block (CCB) : 0x624D9560
         State of The Call        : STATE_ACTIVE
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.231
         Source IP Port    (Media): 20500
         Destn  IP Address (Media): 166.34.245.230
         Destn  IP Port    (Media): 20644
         Destn SIP Addr (Control) : 166.34.245.230
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.230
*Mar  8 17:38:38: The Call Setup Information is :
        Call Control Block (CCB) : 0x624D9560
         State of The Call        : STATE_DEAD
         TCP Sockets Used         : NO
         Calling Number           : 3660110
         Called Number            : 3660210
         Negotiated Codec         : g711ulaw
         Source IP Address (Media): 166.34.245.231
         Source IP Port    (Media): 20500
         Destn  IP Address (Media): 166.34.245.230
         Destn  IP Port    (Media): 20644
         Destn SIP Addr (Control) : 166.34.245.230
         Destn SIP Port (Control) : 5060
         Destination Name         : 166.34.245.230
*Mar  8 17:38:38: 
        Disconnect Cause (CC)    : 16
        Disconnect Cause (SIP)   : 200
Related Commands

Command

Description

debug ccsip all

Enables all SIP-related debugging.

debug ccsip error

Displays SIP SPI errors. This command traces all error messages generated from errors encountered by the SIP subsystem.

debug ccsip events

Displays all SIP SPI events tracing and traces the events posted to SIP SPI from all interfaces.

debug ccsip messages

Displays all SIP SPI message tracing and traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.

debug ccsip states

Displays all SIP SPI state tracing and traces the state machine changes of SIP SPI and displays the state transitions.

debug ccsip error

To display SIP Service Provider Interface (SPI) errors, enter the debug ccsip error command in EXEC mode.

debug ccsip error
Syntax Description

This command has no arguments or keywords.

Defaults

SIP SPI errors are not displayed

Command Modes

EXEC

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

The output of the command was changed.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

This command traces all error messages generated from errors encountered by the SIP subsystem.

Examples

The following example displays debug output from one side of the call:
Router1# debug ccsip error
SIP Call error tracing is enabled
Router1#
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  act_idle_call_setup
*Mar  6 14:16:41:  act_idle_call_setup:Not using Voice Class Codec
*Mar  6 14:16:41: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160 
*Mar  6 14:16:41: REQUEST CONNECTION TO IP:166.34.245.231 PORT:5060
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  act_idle_connection_created
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  act_idle_connection_created: Connid(1) created to 
166.34.245.231:5060, local_port 55674
*Mar  6 14:16:41: sipSPIAddLocalContact
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  6 14:16:41: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:5060
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  act_sentinvite_new_message
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:16:41:  Roundtrip delay 4 milliseconds for method INVITE
*Mar  6 14:16:41: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:5060
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  act_recdproc_new_message
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  sipSPICheckResponse : Updating session description
*Mar  6 14:16:41: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:16:41:  Roundtrip delay 8 milliseconds for method INVITE
*Mar  6 14:16:41: HandleSIP1xxRinging: SDP MediaTypes negotiation successful!
Negotiated Codec      : g711ulaw , bytes :160
Inband Alerting       : 0 
*Mar  6 14:16:45: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:5060
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  act_recdproc_new_message
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  sipSPICheckResponse : Updating session description
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:16:45:  Roundtrip delay 3844 milliseconds for method INVITE
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  act_recdproc_new_message: SDP MediaTypes negotiation 
successful!
Negotiated Codec      : g711ulaw , bytes :160
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  sipSPIReconnectConnection
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  recv_200_OK_for_invite
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  6 14:16:45: HandleUdpReconnection: Udp socket connected for fd: 1 with 
166.34.245.231:5060
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  ccsip_caps_ind
*Mar  6 14:16:45: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Mar  6 14:16:45: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE
*Mar  6 14:16:45: CCSIP-SPI-CONTROL:  ccsip_caps_ack
*Mar  6 14:16:49: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.231:56101
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  act_active_new_message
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  sact_active_new_message_request
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  sipSPIInitiateCallDisconnect : Initiate call 
disconnect(16) for outgoing call
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  act_disconnecting_disconnect
*Mar  6 14:16:49: CCSIP-SPI-CONTROL:  sipSPICallCleanup
*Mar  6 14:16:49: CLOSE CONNECTION TO CONNID:1
*Mar  6 14:16:49: sipSPIIcpifUpdate :CallState: 4 Playout: 2945 DiscTime:48340988 ConnTime 
48340525
*Mar  6 14:16:49: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote 
port: 5060
The following example displays debug output from the other side of the call:
Router2# debug ccsip error
SIP Call error tracing is enabled
Router2#
*Mar  8 17:42:39: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.230:55674
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  sipSPISipIncomingCall
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  act_idle_new_message
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  sact_idle_new_message_invite
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  8 17:42:39:  sact_idle_new_message_invite:Not Using Voice Class Codec
*Mar  8 17:42:39: sact_idle_new_message_invite: Preferred codec[0] type: g711ulaw Bytes 
:160
*Mar  8 17:42:39: sact_idle_new_message_invite: Media Negotiation successful for an
incoming call
*Mar  8 17:42:39: sact_idle_new_message_invite: Negotiated Codec      : g711ulaw, bytes 
:160
Preferred Codec       : g711ulaw, bytes :160
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:42:39: Num of Contact Locations 1 3660110 166.34.245.230 5060
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  act_recdinvite_proceeding
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  ccsip_caps_ind
*Mar  8 17:42:39: ccsip_caps_ind: codec(negotiated) = 5(Bytes 160)
*Mar  8 17:42:39: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Mar  8 17:42:39: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  ccsip_caps_ack
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  act_recdinvite_alerting
*Mar  8 17:42:39:  180 Ringing with SDP - not likely
*Mar  8 17:42:39: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:42:42: CCSIP-SPI-CONTROL:  act_sentalert_connect
*Mar  8 17:42:42: sipSPIAddLocalContact
*Mar  8 17:42:42: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:42:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.230:55674
*Mar  8 17:42:42: CCSIP-SPI-CONTROL:  act_sentsucc_new_message
*Mar  8 17:42:42: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  act_active_disconnect
*Mar  8 17:42:47: REQUEST CONNECTION TO IP:166.34.245.230 PORT:5060
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  act_active_connection_created
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sipSPICheckSocketConnection
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sipSPICheckSocketConnection: Connid(1) created to 
166.34.245.230:5060, local_port 56101
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sip_stats_method
*Mar  8 17:42:47: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 
166.34.245.230:55674
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  act_disconnecting_new_message
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sact_disconnecting_new_message_response
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sipSPICheckResponse
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sip_stats_status_code
*Mar  8 17:42:47:  Roundtrip delay 0 milliseconds for method BYE
*Mar  8 17:42:47: CCSIP-SPI-CONTROL:  sipSPICallCleanup
*Mar  8 17:42:47: CLOSE CONNECTION TO CONNID:1
*Mar  8 17:42:47: sipSPIIcpifUpdate :CallState: 4 Playout: 1255 DiscTime:66856757 ConnTime 
66856294
*Mar  8 17:42:47: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote 
port: 5060
Related Commands

Command

Description

debug ccsip all

Enables all SIP-related debugging.

debug ccsip calls

Displays all SIP SPI call tracing and traces the SIP call details as they are updated in the SIP call control block.

debug ccsip events

Displays all SIP SPI events tracing and traces the events posted to SIP SPI from all interfaces.

debug ccsip messages

Displays all SIP SPI message tracing and traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.

debug ccsip states

Displays all SIP SPI state tracing and traces the state machine changes of SIP SPI and displays the state transitions.

debug ccsip events

To display all Session Initiation Protocol (SIP) Service Provider Interface (SPI) events tracing and traces the events posted to SIP SPI from all interfaces, enter the debug ccsip events command in EXEC mode.

debug ccsip events
Syntax Description

This command has no arguments or keywords.

Command Modes

EXEC

Defaults

SIP SPI events tracing is not displayed

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

The output of the command was changed.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

This command traces the events posted to SIP SPI from all interfaces.

Examples

The following example shows debug output from one side of the call:
Router1# debug ccsip events
SIP Call events tracing is enabled
Router1#
*Mar  6 14:17:57:  Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Mar  6 14:17:57:  Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
*Mar  6 14:17:57:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  6 14:18:00:  Queued event from SIP SPI : SIPSPI_EV_RECONNECT_CONNECTION
*Mar  6 14:18:00:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  6 14:18:04:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  6 14:18:04:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
*Mar  6 14:18:04:  Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
The following example shows debug output from the other side of the call:
Router2# deb ccsip events
SIP Call events tracing is enabled
Router2#
*Mar  8 17:43:55:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:43:55:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING
*Mar  8 17:43:55:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_ALERTING
*Mar  8 17:43:55:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:43:58:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_CONNECT
*Mar  8 17:43:58:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:44:01:  Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
*Mar  8 17:44:01:  Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
*Mar  8 17:44:01:  Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar  8 17:44:01:  Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
Related Commands

Command

Description

debug ccsip all

Enables all SIP-related debugging.

debug ccsip calls

Displays all SIP SPI call tracing and traces the SIP call details as they are updated in the SIP call control block.

debug ccsip error

Displays SIP SPI errors. This command traces all error messages generated from errors encountered by the SIP subsystem.

debug ccsip messages

Displays all SIP SPI message tracing and traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.

debug ccsip states

Displays all SIP SPI state tracing and traces the state machine changes of SIP SPI and displays the state transitions.

debug ccsip messages

To display all Session Initiation Protocol (SIP) Service Provider Interface (SPI) message tracing and to trace the SIP messages exchanged between the SIP UA client (UAC) and the access server, enter the debug ccsip messages command in Privileged EXEC mod.

debug ccsip messages
Syntax Description

This command has no arguments or keywords.

Defaults

SIP SPI message tracing is not displayed

Command Modes

EXEC

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

The output of the command was changed.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

This command traces the SIP messages exchanged between the SIP user agent client (UAC) and the access server.

Examples

The following example shows debug output from one side of the call:
Router1# debug ccsip messages
SIP Call messages tracing is enabled
Router1#
*Mar  6 14:19:14: Sent: 
INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Sat, 06 Mar 1993 19:19:14 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Cisco-Guid: 2881152943-2184249568-0-483551624
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731427554
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 138
v=0
o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20762 RTP/AVP 0
*Mar  6 14:19:14: Received: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:45:12 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Timestamp: 731427554
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Length: 0
*Mar  6 14:19:14: Received: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:45:12 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Timestamp: 731427554
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 138
v=0
o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20224 RTP/AVP 0
*Mar  6 14:19:16: Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
Date: Mon, 08 Mar 1993 22:45:12 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Timestamp: 731427554
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 138
v=0
o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20224 RTP/AVP 0
*Mar  6 14:19:16: Sent: 
ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
Date: Sat, 06 Mar 1993 19:19:14 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Max-Forwards: 6
Content-Type: application/sdp
Content-Length: 138
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20762 RTP/AVP 0
*Mar  6 14:19:19: Received: 
BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.231:53600
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
To: "3660110" <sip:3660110@166.34.245.230>
Date: Mon, 08 Mar 1993 22:45:14 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 731612717
CSeq: 101 BYE
Content-Length: 0
*Mar  6 14:19:19: Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.231:53600
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
To: "3660110" <sip:3660110@166.34.245.230>
Date: Sat, 06 Mar 1993 19:19:19 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Timestamp: 731612717
Content-Length: 0
CSeq: 101 BYE
The following example show debug output from the other side of the call:
Router2# debug ccsip messages
SIP Call messages tracing is enabled
Router2#
*Mar  8 17:45:12: Received: 
INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Sat, 06 Mar 1993 19:19:14 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Cisco-Guid: 2881152943-2184249568-0-483551624
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731427554
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 138
v=0
o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20762 RTP/AVP 0
*Mar  8 17:45:12: Sent: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:45:12 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Timestamp: 731427554
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Length: 0
*Mar  8 17:45:12: Sent: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:45:12 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Timestamp: 731427554
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 138
v=0
o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20224 RTP/AVP 0
*Mar  8 17:45:14: Sent: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
Date: Mon, 08 Mar 1993 22:45:12 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Timestamp: 731427554
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 138
v=0
o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20224 RTP/AVP 0
*Mar  8 17:45:14: Received: 
ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.230:55820
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
Date: Sat, 06 Mar 1993 19:19:14 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Max-Forwards: 6
Content-Type: application/sdp
Content-Length: 138
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20762 RTP/AVP 0
*Mar  8 17:45:17: Sent: 
BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP  166.34.245.231:53600
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
To: "3660110" <sip:3660110@166.34.245.230>
Date: Mon, 08 Mar 1993 22:45:14 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 731612717
CSeq: 101 BYE
Content-Length: 0
*Mar  8 17:45:17: Received: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP  166.34.245.231:53600
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357
To: "3660110" <sip:3660110@166.34.245.230>
Date: Sat, 06 Mar 1993 19:19:19 GMT
Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Timestamp: 731612717
Content-Length: 0
CSeq: 101 BYE
Related Commands

Command

Description

debug ccsip all

Enables all SIP-related debugging.

debug ccsip calls

Displays all SIP SPI call tracing and traces the SIP call details as they are updated in the SIP call control block.

debug ccsip error

Displays SIP SPI errors. This command traces all error messages generated from errors encountered by the SIP subsystem.

debug ccsip events

Displays all SIP SPI events tracing and traces the events posted to SIP SPI from all interfaces.

debug ccsip states

Displays all SIP SPI state tracing and traces the state machine changes of SIP SPI and displays the state transitions.

debug ccsip states

To display all Session Initiation Protocol (SIP) Service Provider Interface (SPI) state tracing, and to trace the state machine changes of SIP SPI and to display the stat transitions, enter the debug ccsip states command in EXEC mode.

debug ccsip states
Syntax Description

This command has no arguments or keywords.

Defaults

SIP SPI state tracing is not displayed

Command Modes

EXEC

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

This command traces the state machine changes of SIP SPI and displays the state transitions.

Examples

The following example shows output for the debug ccsip states command:
Router# debug ccsip states
SIP Call states tracing is enabled
UA-1#
*Jan 2 18:34:37.793:0x6220C634 :State change from (STATE_NONE, SUBSTATE_NONE) to 
(STATE_IDLE, SUBSTATE_NONE)
*Jan 2 18:34:37.797:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_NONE) to 
(STATE_IDLE, SUBSTATE_CONNECTING)
*Jan 2 18:34:37.797:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to 
(STATE_IDLE, SUBSTATE_CONNECTING)
*Jan 2 18:34:37.801:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to 
(STATE_SENT_INVITE, SUBSTATE_NONE)
*Jan 2 18:34:37.809:0x6220C634 :State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to 
(STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
*Jan 2 18:34:37.853:0x6220C634 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING)
*Jan 2 18:34:38.261:0x6220C634 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE)
*Jan 2 18:35:09.860:0x6220C634 :State change from (STATE_ACTIVE, SUBSTATE_NONE) to 
(STATE_DISCONNECTING, SUBSTATE_NONE)
*Jan 2 18:35:09.868:0x6220C634 :State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to 
(STATE_DEAD, SUBSTATE_NONE)
*Jan 2 18:28:38.404: Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION
Related Commands

Command

Description

debug ccsip all

Enables all SIP-related debugging.

debug ccsip calls

Displays all SIP SPI call tracing and traces the SIP call details as they are updated in the SIP call control block.

debug ccsip error

Displays SIP SPI errors. This command traces all error messages generated from errors encountered by the SIP subsystem.

debug ccsip events

Displays all SIP SPI events tracing and traces the events posted to SIP SPI from all interfaces.

debug ccsip messages

Displays all SIP SPI message tracing and traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.

default

To reset the value of a command to its default, enter the default command in SIP user-agent configuration mode.

default {aaa username | max-forwards | retry {invite | response | bye | cancel} | sip-server | timers {trying | connect | disconnect | expires} | transport {tcp | udp}
Syntax Description

aaa username

Resets AAA related configuration.

max-forwards

Resets max-forwards to its default of 6.

retry {invite | response | bye | cancel}

Resets the specified retry to its default. (6 for invite and response; 10 for bye and cancel).

sip-server

Resets the sip-server to a null value.

timers {trying | connect | disconnect | expires}

Resets the specified retry to its default (500 for trying, connect, and disconnect; 180000 for expires).

transport {tcp | udp}

Turns on the handling of SIP messages via UDP or TCP.

Defaults

No default behavior or values

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Examples

The following example shows how to set max-forwards to its default of 6:
Router(config)# sip-ua
Router(config-sip-ua)# default max-forwards
Related Commands

Command

Description

cap-list vfc

Adds a voice codec overlay file to the capability file list.

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

gw-accounting

To enable Voice over IP (VoIP) gateway-specific accounting and to define the accounting method, use the gw-accounting command in global configuration mode. To disable gateway-specific accounting, use the no form of this command.

gw-accounting {h323 [vsa] | syslog | voip}
no gw-accounting {h323 [vsa] | syslog | voip}
Syntax Description

h323

Enables standard H.323 accounting using Internet Engineering Task Force (IETF) RADIUS attributes.

vsa

(Optional) Enables H.323 accounting using RADIUS vendor-specific attributes (VSAs).

syslog

Enables the system logging facility to output accounting information in the form of a system log message.

voip

Enables generic gateway-specific accounting.

Defaults

Disabled

Command Modes

Global configuration

Command History

Release

Modification

11.3(6)NA2

This command was introduced on the Cisco 2500 and Cisco 3600 series routers and the AS5300 universal access server.

12.0(7)T

The vsa keyword was added.

12.1(1)T

The voip keyword was added.

12.2(2)XA

This command was implemented for the Cisco AS5400 and Cisco AS5350.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T for the Cisco AS5850 universal gateway.

Usage Guidelines

To collect basic start-stop connection accounting data, the gateway must be configured to support gateway-specific H.323 accounting functionality. The gw-accounting command enables you to send accounting data to the RADIUS server in one of four ways:

•Using standard IETF RADIUS accounting attribute/value (AV) pairs—This method is the basic method of gathering accounting data (connection accounting) according to the specifications defined by the IETF. Use the gw-accounting h323 command to configure the standard IETF RADIUS method of applying H.323 gateway-specific accounting. Table 2 shows the supported IETF RADIUS attributes.

Table 2 Supported IETF RADIUS Accounting Attributes

Number

Attribute

Description

30

Called-Station-Id

Allows the network access server to send the telephone number that the user called as part of the Access-Request packet (using Dialed Number Identification Service [DNIS] or similar technology). This attribute is supported only on ISDN and modem calls on the Cisco AS5200 and Cisco AS5300 universal access server if used with ISDN PRI.

31

Calling-Station-Id

Allows the network access server to send the telephone number that the call came from as part of the Access-Request packet (using Automatic Number Identification or similar technology). This attribute has the same value as "remote-addr" from TACACS+. This attribute is supported only on ISDN, and modem calls on the Cisco AS5200 and Cisco AS5300 universal access server if used with ISDN PRI.

42

Acct-Input-Octets

Indicates how many octets have been received from the port over the course of the accounting service being provided.

43

Acct-Output-Octets

Indicates how many octets have been sent to the port over the course of delivering the accounting service.

44

Acct-Session-Id

Indicates a unique accounting identifier that makes it easy to match start and stop records in a log file. Acct-Session-Id numbers restart at 1 each time the router is power-cycled or the software is reloaded.

47

Acct-Input-Packets

Indicates how many packets have been received from the port over the course of this service being provided to a framed user.

48

Acct-Output-Packets

Indicates how many packets have been sent to the port in the course of delivering this service to a framed user.

For more information about RADIUS and the use of IETF-defined attributes, refer to the Cisco IOS Security Configuration Guide.

•Overloading the Acct-Session-Id field—Attributes that cannot be mapped to standard RADIUS are packed into the Acct-Session-Id attribute field as ASCII strings separated by the character "/". The Acct-Session-Id attribute is defined to contain the RADIUS account session ID, which is a unique identifier that links accounting records associated with the same login session for a user. To support additional fields, we have defined the following string format for this field:
<session id>/<call leg setup time>/<gateway id>/<connection id>/<call origin>/ 
<call type>/<connect time>/<disconnect time>/<disconnect cause>/<remote ip address>
Table 3 shows the field attributes that you use with the overloaded session-ID method and a brief description of each.

Table 3 Field Attributes in Overloaded Acct-Session-ID

Field Attribute

Description

Session-Id

Specifies the standard RADIUS account session ID.

Setup-Time

Provides the Q.931 setup time for this connection in Network Time Protocol (NTP) format. NTP time formats are displayed as %H: %M: %S %k %Z %tw %tn %td %Y where:

%H is hour (00 to 23).

%M is minutes (00 to 59).

%S is seconds (00 to 59).

%k is milliseconds (000 to 999).

%Z is timezone string.

%tw is day of week (Saturday through Sunday).

%tn is month name (January through December).

%td is day of month (01 to 31).

%Y is year including century (for example, 1998).

Gateway-Id

Indicates the name of the underlying gateway in the form "gateway.domain_name."

Call-Origin

Indicates the origin of the call relative to the gateway. Possible values are originate and answer.

Call-Type

Indicates the call leg type. Possible values are telephony and VoIP.

Connection-Id

Specifies the unique global identifier used to correlate call legs that belong to the same end-to-end call. The field consists of 4 long words (128 bits). Each long word is displayed as a hexadecimal value and is separated by a space character.

Connect-Time

Provides the Q.931 connect time for this call leg, in NTP format.

Disconnect-Time

Provides the Q.931 disconnect time for this call leg, in NTP format.

Disconnect-Cause

Specifies the reason a call was taken offline as defined in the Q.931 specification.

Remote-Ip-Address

Indicates the address of the remote gateway port where the call is connected.

Because of the limited size of the Acct-Session-Id string, it is not possible to embed very many information elements in it. Therefore, this feature supports only a limited set of accounting information elements.

Use the gw-accounting h323 command to configure the overloaded session ID method of applying H.323 gateway-specific accounting.

•Using vendor-specific RADIUS attributes—The IETF draft standard specifies a method for communicating vendor-specific information between the network access server and the RADIUS server by using the vendor-specific attribute (Attribute 26). Vendor-specific attributes (VSAs) allow vendors to support their own extended attributes not suitable for general use. The Cisco RADIUS implementation supports one vendor-specific option using the format recommended in the specification. The Cisco vendor ID is 9, and the supported option has vendor-type 1, which is named "cisco-avpair." The value is a string of the format:
protocol: attribute sep value *
"Protocol" is a value of the Cisco "protocol" attribute for a particular type of authorization. "Attribute" and "value" are an appropriate attribute/value (AV) pair defined in the Cisco TACACS+ specification, and "sep" is "=" for mandatory attributes and "*" for optional attributes. This allows the full set of features available for TACACS+ authorization to also be used for RADIUS.

The VSA fields and their ASCII values are listed in Table 4.

Table 4 VSA Fields and Their ASCII Values

IETF RADIUS Attribute

Vendor-
Specific Company Code

Subtype Number

Attribute Name

Description

26

9

23

h323-remote-address

Indicates the IP address of the remote gateway.

26

9

24

h323-conf-id

Identifies the conference ID.

26

9

25

h323-setup-time

Indicates the setup time for this connection in Coordinated Universal Time (UTC) formerly known as Greenwich Mean Time (GMT) and Zulu time.

26

9

26

h323-call-origin

Indicates the origin of the call relative to the gateway. Possible values are originating and terminating (answer).

26

9

27

h323-call-type

Indicates the call leg type. Possible values are telephony and VoIP.

26

9

28

h323-connect-time

Indicates the connection time for this call leg in UTC.

26

9

29

h323-disconnect-time

Indicates the time this call leg was disconnected in UTC.

26

9

30

h323-disconnect-cause

Specifies the reason a connection was taken offline per the Q.931 specification.

26

9

31

h323-voice-quality

Specifies the impairment factor (ICPIF) affecting voice quality for a call.

26

9

33

h323-gw-id

Indicates the name of the underlying gateway.

Use the gw-accounting h323 vsa command to configure the VSA method of applying H.323 gateway-specific accounting.

•Using syslog records—The syslog accounting option exports the information elements associated with each call leg through a system log message, which can be captured by a syslog daemon on the network. The syslog output consists of the following:
<server timestamp> <gateway id> <message number> : <message label> : <list of AV 
pairs>
The syslog message fields are listed in Table 5.

Table 5 Syslog Mesage Output Fields

Field

Description

server timestamp

The time stamp created by the server when it receives the message to log.

gateway id

The name of the gateway that emits the message.

message number

The number assigned to the message by the gateway.

message label

A string used to identify the message category.

list of AV pairs

A string that consists of <attribute name> <attribute value> pairs
separated by commas.

Use the gw-accounting syslog command to configure the syslog record method of gathering H.323 accounting data.

Use this command if you configure the AAA accounting application.

If you enable both h323 and syslog simultaneously, CDRs are generated in both methods.

Examples

The following example configures basic H.323 accounting using IETF RADIUS attributes:
gw-accounting h323
The following example configures H.323 accounting using VSA RADIUS attributes:
gw-accounting h323 vsa
The following example enables gateway-specific accounting and defines the accounting method as voip:
gw-accounting voip
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

max-forwards

To set the maximum number of proxy or redirect servers that can forward the request, use the max-forwards command in SIP user agent configuration mode. To restore the default value, use the no form of this command.

max-forwards number
no max-forwards
Syntax Description

number

Number of hops. Possible values are 1 through 15. The default is 6.

Defaults

6

Command Modes

SIP user agent configuration

Command History

Release

Modification

12.1(3)T

This command was introduced on the Cisco 2600 and Cisco 3600 series routers and on the Cisco AS5300 universal access server.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was introduced on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command does not support the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

To reset this command to the default value, you can also use the default command.

Examples

The following is an example of setting the number of forwarding requests to proxy or redirect servers:
sip-ua
 max-forwards 2
Related Commands

Command

Description

max-redirects

Sets the maximum number of redirects that the user agent allows.

max-redirects

To set the maximum number of redirect servers that the user agent allows, use the max-redirects command in dial-peer configuration mode. To restore the default value, use the no form of this command.

max-redirects number
no max-redirects
Syntax Description

number

Maximum number of redirect servers that a call can traverse. Range is 1 to 10. The default is 1.

Defaults

1

Command Modes

Dial-peer configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on the Cisco 2600 and Cisco 3600 series routers and on the Cisco AS5300 universal access server.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series. This command does not support the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Examples

The following is an example of setting the maximum number of redirect servers that the user agent allows:
dial-peer voice 102 voip
 max-redirects 2
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

retry bye

To configure the number of times that a BYE request is retransmitted to the other user agent, use the retry bye command in SIP user-agent configuration mode. To reset to the default, use the no form of this command.

retry bye number
no retry bye number
Syntax Description

number

Number of BYE retries. Range is 1 to 10. The default is 10.

Defaults

10 retries

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on Cisco 2600 and Cisco 3600 series routers and Cisco AS5300 universal access servers.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on Cisco AS5850 universal gateways.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command was not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

To reset this command to the default value, you can also use the default command.

Examples
In the following example, the number of BYE retries has been set to 5.
sip-ua
 retry bye 5
Related Commands

Command

Description

default

Resets the value of a command to its default.

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

retry cancel

To configure the number of times that a CANCEL request is retransmitted to the other user agent, use the retry cancel command in SIP user-agent configuration mode. To reset to the default, use the no form of this command.

retry cancel number
no retry cancel number
Syntax Description

number

Number of CANCEL retries. Range is 1 to 10. The default is 10.

Defaults

10 retries

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on Cisco 2600 and Cisco 3600 series routers and Cisco AS5300 universal access servers.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on Cisco AS5850 universal gateways.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command was not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

To reset this command to the default value, you can also use the default command.

Examples
In the following example, the number of cancel retries has been set to 5.
sip-ua
 retry cancel 5
Related Commands

Command

Description

default

Resets the value of a command to its default.

sip-ua

Enables the sip-ua configuration commands, with which you configure the user agent.

retry invite

To configure the number of times that a Session Initiation Protocol (SIP) INVITE request is retransmitted to the other user agent, use the retry invite command in SIP user-agent configuration mode. To reset to the default, use the no form of this command.

retry invite number
no retry invite number
Syntax Description

number

Number of INVITE retries. Range is 1 to 10. The default is 6.

Defaults

6 retries

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on Cisco 2600 and Cisco 3600 series routers and Cisco AS5300 universal access servers.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on Cisco AS5850 universal gateways.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command was not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

To reset this command to the default value, you can also use the default command.

When configuring SIP using SIP user-agent configuration commands such as the retry invite command, the use of the default values for the commands causes the rotary function to not take effect. The rotary function is when you set up more than one VoIP dial peer for the same destination pattern, and the dial peers are assigned to different targets. The purpose of assigning different targets is if the call cannot be set up with the first dial peer (preference one), the next dial peer can be tried. For example:
dial-peer voice 201 voip
 destination-pattern 1234567
 codec g711ulaw
 session protocol sipv2
 session target ipv4:1.2.3.4
dial-peer voice 202 voip
 destination-pattern 12345..
 codec g711ulaw
 session protocol sipv2
 session target ipv4:10.2.3.42
To use the rotary function within SIP, set the retry value for the SIP retry invite command to 4 or less. For example:
sip-ua
retry invite 4
Examples
In the following example, the number of invite retries is set to 5.
sip-ua
 retry invite 5
Related Commands

Command

Description

default

Resets the value of a command to its default.

sip-ua

Enables the sip-ua configuration commands, with which you configure the user agent.

retry response

To configure the number of times that the RESPONSE message is retransmitted to the other user agent, use the retry response command in SIP user-agent configuration mode. To reset to the default, use the no form of this command.

retry response number
no retry response number
Syntax Description

number

Number of RESPONSE retries. Range is 1 to 10. Default is 6.

Defaults

6 retries

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on Cisco 2600 and Cisco 3600 series routers and Cisco AS5300 universal access servers.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on Cisco AS5850 universal gateways.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command was not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

To reset this command to the default value, you can also use the default command.

Examples
The following example sets the number of response retries to 5.
sip-ua
 retry response 5
Related Commands

Command

Description

default

Resets the value of a command to its default.

sip-ua

Enables the sip-ua configuration commands, with which you configure the user agent.

session protocol

To specify a session protocol for calls between local and remote routers using the packet network, use the session protocol command in dial-peer configuration mode. To reset to the default, use the no form of this command.

session protocol {aal2-trunk | cisco | sipv2 | smtp}
no session protocol
Syntax Description

aal2-trunk

Dial peer uses ATM adaptation layer 2 (AAL2) nonswitched trunk session protocol.

cisco

Dial peer uses the proprietary Cisco Voice over IP (VoIP) session protocol.

sipv2

Dial peer uses the Internet Engineering Task Force (IETF) Session Initiation Protocol (SIP). Use this keyword with the SIP option.

smtp

Dial peer uses Simple Mail Transfer Protocol (SMTP) session protocol.

Defaults

No default behaviors or values

Command Modes

Dial-peer configuration

Command History

Release

Modification

11.3(1)T

This command was introduced for VoIP peers on Cisco 3600 series routers.

12.0(3)XG

Support was added for Voice over Frame Relay (VoFR) dial peers.

12.0(4)XJ

This command was modified for store-and-forward fax on Cisco AS5300 universal access servers.

12.1(1)XA

This command was implemented for VoATM dial peers on Cisco MC3810 multiservice access concentrators, and the aal2-trunk keyword was added.

12.1(1)T

This command was integrated into Cisco IOS Release 12.1(1)T, and the sipv2 keyword was added.

12.1(2)T

This command was integrated into Cisco IOS Release 12.1(2)T.

12.2(2)T

This command was implemented on Cisco 7200 series routers.

12.2(4)T

This command was introduced on Cisco 1750 access routers.

12.2(8)T

This command was introduced on the Cisco 1751, Cisco 2600, Cisco 3600, Cisco 3725,and Cisco 3745 platforms.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command is not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

Note The aal2-trunk and smtp keywords are not supported on Cisco 7200 series routers.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

The cisco keyword is applicable only to VoIP on the Cisco 1750, Cisco 1751, Cisco 3600 series, and Cisco 7200 series routers.

The aal2-trunk keyword is applicable only to VoATM on the Cisco MC3810 multiservice access concentrator and the Cisco 7200 series router.

This command applies to both on-ramp and off-ramp store-and-forward fax functions.

Examples

The following example shows that AAL2 trunking has been configured as the session protocol:
dial-peer voice 10 voatm
 session protocol aal2-trunk
The following example shows that Cisco session protocol has been configured as the session protocol:
dial-peer voice 20 voip
 session protocol cisco
The following example shows that a VoIP dial peer for SIP has been configured as the session protocol for VoIP call signaling:
dial-peer voice 102 voip
 session protocol sipv2
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.

session target (VoIP)

Configures a network-specific address for a dial peer.

session target (VoIP)

To specify a network-specific address for a dial peer, use the session target command in dial-peer configuration mode. To restore default values for this parameter, use the no form of this command.

Note This command applies to all dial peers except plain old telephone service (POTS) dial peers.

session target {sip-server | ipv4:destination-address[:port-number] | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp | loopback:compressed | loopback:uncompressed | ras | settlement provider-number}
no session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.]
host-name | loopback:rtp | loopback:compressed | loopback:uncompressed | ras | settlement provider-number}
Syntax Description

sip-server

Sets the session target to the global SIP server.

ipv4:destination-address

Sets the IP address of the dial peer.

port-number

(Optional) Specifies the port number to contact to complete the call leg.

dns:[$s$...]host-name

Indicates that the domain name server resolves the name of the IP address. Valid entries for this parameter are characters representing the name of the host device.

(Optional) Use one of the following three wildcards with this keyword when defining the session target for Voice over IP (VoIP) peers:

•$s$.—Indicates that the source destination pattern will be used as part of the domain name.

•$d$.—Indicates that the destination number will be used as part of the domain name.

•$e$.—Indicates that the digits in the called number will be reversed, periods will be added between the digits of the called number, and this string will be used as part of the domain name.

•$u$.—Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name.

loopback:rtp

Indicates that all voice data will be looped back to the source. This is applicable for VoIP peers.

loopback:compressed

Indicates that all voice data will be looped back in compressed mode to the source. This is applicable for POTS peers.

loopback:uncompressed

Indicates that all voice data will be looped back in uncompressed mode to the source. This is applicable for POTS peers.

ras

Indicates that the registration, admission, and status (RAS) signaling function protocol is being used, meaning that a gatekeeper will be consulted to translate the E.164 address into an IP address.

settlement provider-number

Indicates that the settlement server is the target to resolve the terminating gateway address. Enter the provider IP address for provider- number.

Defaults

The default for this command is enabled with no IP address or domain name defined.

Command Modes

Dial-peer configuration

Command History

Release

Modification

11.3(1)T

This command was introduced.

11.3(1)MA

Support was added for VoFR, VoATM, VoHDLC, and POTS dial peers on the Cisco MC3810 multiservice concentrator.

12.0(3)T

Support was added for VoIP and POTS dial peers on the Cisco AS5300 universal access server. The parameter was added for RAS.

12.0(3)XG

Support was added for VoFR dial peers on the Cisco 2600 series and 3600 series routers.

12.0(4)T

Support was added for VoFR and POTS dial peers on the Cisco 7200 series routers, and the support added in Cisco IOS Release 12.0(3)XG was integrated into Cisco IOS Release 12.0(4)T.

12.0(4)XJ

Support was added for store-and-forward fax on the Cisco AS5300 universal access server platform.

12.1(1)T

The sip-server option was added.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

Enter the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.

You can enter the session target dns command with or without the specified wild cards. Using the optional wildcards can reduce the number of VoIP dial-peer session targets you need to configure if you have groups of numbers associated with a particular router.

Examples

The following example shows how to configure a session target using DNS for a host, "voice_router," in the domain "cisco.com":
Router(config)# dial-peer voice 102 voip
Router(config-dial-peer)# session target dns:voice_router.cisco.com
The following example shows how to configure a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number—in this case, the four-digit extension—to identify the dial peer. As in the preceding example, the domain is "cisco.com."
Router(config)# dial-peer voice 10 voip
Router(config-dial-peer)# destination-pattern 1310222....
Router(dial-peer-config)# session target dns:$u$.cisco.com
The following example shows how to configure a session target using DNS, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
Router(config)# dial-peer voice 10 voip
Router(config-dial-peer)# destination-pattern 13102221111
Router(config-dial-peer)# session target dns:$d$.cisco.com
The following example shows how to configure a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
Router(config)# dial-peer voice 10 voip
Router(config-dial-peer)# destination-pattern 12345
Router(config-dial-peer)# session target dns:$e$.cisco.com
The following example shows how to configure a session target using RAS:
Router(config)# dial-peer voice 11 voip
Router(config-dial-peer)# destination-pattern 13102221111
Router(config-dial-peer)# session target ras
The following example shows how to configure a session target using the settlement server:
Router(dial-peer-config)# session target settlement:0
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

sip-server

Configures the SIP server interface.

session transport

To configure the VoIP dial peer to use TCP or User Datagram Protocol (UDP) as the underlying transport layer protocol for Session Initiation Protocol (SIP) messages, use the session transport command in dial-peer configuration mode. To reset this command to the default value, use the no form of this command.

session transport {udp | tcp}
no session transport
Syntax Description

udp

Configure the SIP dial peer to use the UDP transport layer protocol. This is the default.

tcp

Configure the SIP dial peer to use the TCP transport layer protocol.

Defaults

The SIP dial peer uses UDP.

Note The transport protocol for transport and session transport must be the same.

Command Modes

Dial-peer configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

Use the show sip-ua status command to ensure that the transport protocol that you set using the session transport command matches the protocol set using the transport command.

Examples

The following example shows how to configure a VoIP dial peer to use UDP as the underlying transport layer protocol for SIP messages:
Router(config)# dial-peer voice 102 voip
Router(dial-peer-config)# session transport udp
Related Commands

Command

Description

dial-peer voice

Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation.

show sip-ua

Displays information and settings for the Session Initiation Protocol (SIP) user agent (UA)

transport

Configures the Session Initiation Protocol (SIP) user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.

show sip-ua

To display information and settings for the Session Initiation Protocol (SIP) user agent (UA), use the show sip-ua command in privileged EXEC mode.

show sip-ua {map {pstn-sip | sip-pstn} | retry | statistics | status | timers}
Syntax Description

map

Displays the PSTN cause to SIP status code or SIP status code to PSTN cause code mapping table.

pstn-sip

(Optional) Displays the PSTN cause to SIP status code mapping table.

sip-pstn

(Optional) Display the SIP status code to PSTN cause mapping table

retry

Displays SIP protocol retry counts.

statistics

Displays SIP UA response, traffic, and retry statistics.

status

Displays SIP UA listener status.

timers

Displays SIP UA protocol timers.

Defaults

No default behavior or values

Command Modes

Privileged EXEC

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

The following changes were made:

•The statistics keyword was added.

•The statistics portion of the output from the status keyword was moved from the status keyword to the statistics keyword.

•The output from the timers keyword was changed to reflect the changes in the timers command.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Examples

The following example displays output for the show sip-ua retry command:
Router# show sip-ua retry
SIP UA Retry Values
invite retry count = 2
response retry count = 2
bye retry count    = 2
cancel retry count   = 1
The following example displays output for the show sip-ua statistics command:
Router# show sip-ua statistics
SIP Response Statistics (Inbound/Outbound)
    Informational:
      Trying 0/0, Ringing 0/0,
      Forwarded 0/0, Queued 0/0,
      SessionProgress 0/0
    Success:
       OkInvite 0/0, OkBye 0/0,
       OkCancel 0/0, OkOptions 0/0
    Redirection (Inbound only):
      MultipleChoice 0, MovedPermanently 0,
      MovedTemporarily 0, SeeOther 0,
      UseProxy 0, AlternateService 0
    Client Error:
      BadRequest 0/0, Unauthorized 0/0,
      PaymentRequired 0/0, Forbidden 0/0,
      NotFound 0/0, MethodNotAllowed 0/0,
      NotAcceptable 0/0, ProxyAuthReqd 0/0,
      ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
      LengthRequired 0/0, ReqEntityTooLarge 0/0,
      ReqURITooLarge 0/0, UnsupportedMediaType 0/0,
      BadExtension 0/0, TempNotAvailable 0/0,
      CallLegNonExistent 0/0, LoopDetected 0/0,
      TooManyHops 0/0, AddrIncomplete 0/0,
      Ambiguous 0/0, BusyHere 0/0
    Server Error:
      InternalError 0/0, NotImplemented 0/0,
      BadGateway 0/0, ServiceUnavail 0/0,
      GatewayTimeout 0/0, BadSipVer 0/0
    Global Failure:
      BusyEverywhere 0/0, Decline 0/0,
      NoExistAnywhere 0/0, NotAcceptable 0/0
SIP Total Traffic Statistics (Inbound/Outbound)
    Invite 0/0, Ack 0/0, Bye 0/0,
    Cancel 0/0, Options 0/0
Retry Statistics
    Invite 0, Bye 0, Cancel 0, Response 0
The following example displays output for the show sip-ua status command:
Router# show sip-ua status
SIP User Agent Status
SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
SIP max-forwards :6
The following example displays output for the show sip-ua timers command:
Router# show sip-ua timers
SIP UA Timer Values (millisecs)
trying 500, expires 180000, connect 500, disconnect 500
The following example displays output for the show sip-ua map pstn-sip command:
Router# show sip-ua map pstn-sip
The PSTN Cause to SIP Status code mapping table:-
PSTN-Cause   Configured        Default
             SIP-Status       SIP-Status
1               404             404
2               404             404
3               404             404
4               500             500
5               500             500
6               500             500
7               500             500
8               500             500
9               500             500
15              500             500
16              500             500
17              486             486
18              480             480
19              480             480
20              480             480
21              403             403
22              410             410
26              404             404
27              404             404
.
.
.
The following example displays output for the show sip-ua map sip-pstn command:
doc-7204vxr# show sip-ua map sip-pstn
The SIP Status code to PSTN Cause mapping table:-
SIP-Status   Configured        Default
             PSTN-Cause      PSTN-Cause
400             127             127
401             57              57
402             21              21
403             57              57
404             1               1
405             127             127
406             127             127
407             21              21
408             102             102
409             41              41
410             1               1
411             127             127
413             127             127
414             127             127
415             79              79
420             127             127
480             18              18
481             127             127
.
.
.
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

sip-server

To configure a network address for the Session Initiation Protocol (SIP) server interface, use the sip-server command in SIP user-agent configuration mode.

sip-server {dns:[host-name] | ipv4:ip-addr[:port-num]}
Syntax Description

dns:

Sets the global SIP server interface to a Domain Name System (DNS) host name. If you do not specify a host name, the default DNS defined by the ip name-server command is used.

host-name

(Optional) A valid DNS host name in the following format: name.gateway.xyz.

ipv4:ip-addr

Sets the global SIP server interface to an IP address. A valid IP address takes the following format: xxx.xxx.xxx.xxx.

:port-num

(Optional) Specifies the port number for the SIP server.

Defaults

The default for this command is a null value.

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on Cisco 2600 and Cisco 3600 series routers and Cisco AS5300 universal access servers.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on Cisco AS5850 universal gateways.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command is not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

If you use this command, you can then use the session target sip-server command on each dial peer instead of repeatedly entering the SIP server interface address for each dial peer.

To reset this command to a null value, use the default command. This command does not have a no form.

Examples

The following example, beginning in global configuration mode, sets the global SIP server interface to the DNS host name "UA-1-f0.sip.com":
sip-ua
 sip-server dns:UA-1-f0.sip.com
Related Commands

Command

Description

ip name-server

Specifies the address of one or more name servers to use for name and address resolution.

session target (VoIP)

Specifies a network-specific address for a dial peer.

sip-ua

Enters SIP user-agent configuration mode, in order to configure the SIP user agent.

sip-ua

To enable the Session Initiation Protocol (SIP) user-agent configuration commands, in order to configure the user agent, use the sip-ua command in global configuration mode. To reset all SIP user-agent configuration commands to their default values, use the no form of this command.

sip-ua
no sip-ua
Syntax Description

This command has no arguments or keywords.

Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release

Modification

12.1(1)T

This command was introduced on the Cisco 2600, Cisco 3600, and Cisco AS5300 platforms.

12.2(2)XA

This command was implemented on Cisco AS5350 and Cisco AS5400 universal gateways.

12.2(2)XB1

This command was implemented on Cisco AS5850 universal gateways.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command is not supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 platforms in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

Use the sip-ua command to enter SIP user-agent configuration mode. Table 6 lists the SIP user-agent configuration mode commands:

Table 6 SIP User-Agent Configuration Mode Commands

Command

Description

exit

Exits SIP user-agent configuration mode.

inband-alerting

This command is no longer supported as of Cisco IOS Release 12.2. This command is no longer needed because the gateway handles remote or local ringback on the basis of SIP messaging.

max-forwards

Specifies the maximum number of hops for a request.

retry

Configures the SIP signaling timers for retry attempts.

sip-server

Configures a SIP server interface.

timers

Configures the SIP signaling timers.

transport

Enables or disables a SIP user agent transport for TCP or UDP that the protocol SIP user agents listen for on port 5060 (default).

Examples

The following example, beginning in global configuration mode, enters SIP user-agent configuration mode, configures the SIP user agent, and then returns to global configuration mode:
sip-ua
 retry invite 2
 retry response 2
 retry bye 2
 retry cancel 2
 sip-server ipv4:10.0.2.254
 timers invite-wait-100 500
 exit
Related Commands

Command

Description

exit

Exits SIP user-agent configuration mode.

max-forwards

Specifies the maximum number of hops for a request.

retry

Configures the retry attempts for SIP messages.

show sip-ua

Displays statistics for SIP retries, timers, and current listener status.

sip-server

Configures the SIP server interface.

timers

Configures the SIP signaling timers.

transport

Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.

timers

To configure the Session Initiation Protocol (SIP) signaling timers, use the timers command in the Session Initiation Protocol (SIP) user-agent configuration mode. To restore the default value, use the no form of this command.

timers {trying number | connect number | disconnect number | expires number}
no timers {trying number | connect number | disconnect number | expires number}
Syntax Description

trying number

Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 to 1000. The default is 500.

connect number

Time (in milliseconds) to wait for a 200 response to an ACK request. Possible values are 100 to 1000. The default is 500.

disconnect number

Time (in milliseconds) to wait for a 200 response to a BYE request. Possible values are 100 to 1000. The default is 500.

expires number

Time (in milliseconds) for which an INVITE request is valid. Possible values are 60000 to 300000. The default is 180000.

Defaults

trying, connect, and disconnect—500
expires—180000

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

This command was modified to change the names of the parameters. Two of the parameters (invite-wait-180 and invite-wait-200) were combined into one (trying).

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

If you used the previous version of this command to configure timers, your previous timer settings will be maintained. The output of the show running configuration command will reflect both timers.

To reset this command to the default value, you can also use the default command.

Examples

The following example shows how to configure the SIP signaling timers to wait 500 milliseconds for a 100 response to an INVITE request:
Router(config)# sip-ua
Router(config-sip-ua)# timers trying 500 
Related Commands

Command

Description

sip-ua

Enables the SIP user-agent configuration commands, with which you configure the user agent.

transport

To configure the Session Initiation Protocol (SIP) user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket, use the transport command in SIP user-agent configuration mode. To block reception of SIP signaling messages on a particular socket, use the no form of this command.

transport {udp | tcp}
no transport {udp | tcp}
Syntax Description

udp

Configures the SIP user agent to receive SIP messages on UDP port 5060.

tcp

Configures the SIP user agent to receive SIP messages on TCP port 5060.

Defaults

Bboth UDP and TCP transport layer protocols are enabled

Command Modes

SIP user-agent configuration

Command History

Release

Modification

12.1(1)T

This command was introduced.

12.1(3)T

Support for access platforms was added.

12.2(2)XA

Support was added for the Cisco AS5400 and AS5350 universal gateways.

12.2(2)XB1

This command was implemented on the Cisco AS5850 universal gateway.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300 universal access server, Cisco AS5350, Cisco AS5400, and Cisco AS5850 universal gateway is not included in this release.

12.2(11)T

This command was integrated Cisco IOS Release 12.2(11)T on the Cisco AS5850 universal gateway.

Usage Guidelines

This command controls whether messages reach the SIP service provider interface (SPI). By setting udp or tcp as the protocol, this will be the protocol SIP user agents will be listening for on port 5060 (default). To block reception of SIP signaling messages on a specific socket, use the no form of this command.

To reset this command to the default value, use the default command.

Examples

The following example shows how to configure the SIP user agent to block reception of SIP signaling messages on the TCP socket:
Router(config)# sip-ua
Router(config-sip-ua)# no transport tcp
Related Commands

Command

Descr

Hierarchical Navigation

Cisco IOS Software Releases 12.2 T

Session Initiation Protocol (SIP) for VoIP

Downloads

Table Of Contents

Session Initiation Protocol (SIP) for VoIP

Feature Overview

Benefits

Restrictions

Related Features and Technologies

Related Documents

Supported Platforms

Supported Standards, MIBs, and RFCs

Prerequisites

Configuration Tasks

Configuring the SIP User Agent (UA)

Changing the Configuration of the SIP User Agent (UA)

Configuring SIP Support for VoIP Dial Peers

Configuring a POTS Dial Peer

Configuring SIP Call Transfer for a POTS Dial Peer

Configuring SIP Call Transfer for a VoIP Dial Peer

Configuring Phone Number Translation Rules

Verifying the SIP Feature Configuration

Configuration Examples

Basic SIP Configuration Example

Configuring SIP with Multiple Codecs Example

Configuring Phone Number Translation Rules Examples

Call Transfer Configuration Examples

Command Reference

aaa username

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccsip all

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccsip calls

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccsip error

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccsip events

Syntax Description

Command Modes

Defaults

Command History

Usage Guidelines

Examples

Related Commands

debug ccsip messages

Syntax Description

Defaults

Command Modes

Command History

Usage Guidelines

Examples

Related Commands

debug ccsip states

Syntax Description

Defaults