Cisco IOS Software Releases 12.2 T

Table Of Contents

SIP: ISDN Suspend/Resume Support

Contents

Prerequisites for SIP: ISDN Suspend/Resume Support

Information About SIP: ISDN Suspend/Resume Support

Suspend and Resume Overview

SIP Call-Hold Process

How to Configure Suspend and Resume Support

Configuring Suspend and Resume

Troubleshooting Tips

Configuration Examples for SIP: ISDN Suspend/Resume Support

SIP: ISDN Suspend/Resume Support Example

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Command Reference

suspend-resume (SIP)

show sip-ua status

Glossary

SIP: ISDN Suspend/Resume Support

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The SIP: ISDN Suspend/Resume Support feature adds Session Initiation Protocol (SIP) call-hold support to SIP gateways when an ISDN Suspend event is triggered. Because Suspend and Resume support already exists for H.323, the SIP implementation of Suspend and Resume provides feature parity.

Feature Specifications for SIP: ISDN Suspend/Resume Support

Feature History
Release	Modification
12.2(15)T	This feature was introduced.
Supported Platforms
Cisco 1751, Cisco 2610-Cisco 2613, Cisco 2620, Cisco 2621, Cisco 2650, Cisco 2651, Cisco 2691, Cisco 3620, Cisco 3631, Cisco 3640, Cisco 3660, Cisco 3725, Cisco 3745, Cisco 7200 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5850, Cisco CVA120 series, and Cisco uBR925.

Finding Support Information for Platforms and Cisco IOS Software Images

Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.

Contents

•Prerequisites for SIP: ISDN Suspend/Resume Support

•Information About SIP: ISDN Suspend/Resume Support

•How to Configure Suspend and Resume Support

•Configuration Examples for SIP: ISDN Suspend/Resume Support

•Additional References

•Command Reference

•Glossary

Prerequisites for SIP: ISDN Suspend/Resume Support

The following are general prerequisites for SIP deployment:

•Ensure that the gateway has voice functionality that is configurable for SIP.

•Establish a working IP network.

For more information about configuring IP, refer to the following document:

Cisco IOS IP Configuration Guide, Release 12.2

•Configure VoIP.

For more information about configuring VoIP, refer to the following document:

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

Information About SIP: ISDN Suspend/Resume Support

To configure the SIP: ISDN Suspend/Resume Support feature, you need to understand the following concepts:

•Suspend and Resume Overview

•SIP Call-Hold Process

Suspend and Resume Overview

Suspend and Resume are basic functions of ISDN and ISDN User Part (ISUP) signaling procedures and now are a part of SIP functionality. Suspend is described in ITU Q.764 as a message that indicates a temporary cessation of communication that does not release the call. A Suspend message can be accepted during a conversation. A Resume message is received after a Suspend message and is described in ITU Q.764 as a message that indicates a request to recommence communication. If the calling party requests to release the call, the Suspend and Resume sequence is overridden.

SIP Call-Hold Process

When a SIP originating gateway receives an ISDN Suspend message, the originating gateway informs the terminating gateway that there is a temporary cessation of media; that is, the call is placed on hold. There are two ways that SIP gateways receive notice of a call hold. The first way is for the originating gateway to use a connection IP address of 0.0.0.0 (c=0.0.0.0) in the Session Description Protocol (SDP). The information in the SDP is sent in a re-Invite to the terminating gateway. The second way is for the originating gateway to use a=sendonly in the SDP of a re-Invite.

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Note For a SIP gateway to initiate call hold, the c=0.0.0.0 method must be used.

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The purpose of the c=0.0.0.0 line is to notify the terminating gateway to stop sending media packets. When the hold is cancelled and communication is to resume, an ISDN Resume message is sent. The SIP originating gateway takes the call off hold by sending out a re-Invite with the actual IP address of the remote SIP entity in the c= line (in place of 0.0.0.0).

Multiple media fields (m-lines) in the SDP of a re-Invite message are used to indicate media forking, with each m-line representing one media destination. SIP gateways negotiate multiple media streams by using multiple m- and/or c-lines. When an originating gateway receives an ISDN Suspend on a gateway that has negotiated multiple media streams, all of the media streams are placed on hold. The originating gateway sends out a re-Invite that has a c= line that advertises the IP address as 0.0.0.0 on all streams. The originating gateway also mutes the SIP calls for each media stream so that no media is sent to the terminating gateway. When the originating gateway receives an ISDN Resume, it initiates a re-Invite with the original SDP and takes the call off hold.

If the media inactivity timer is configured on the network, the timer is stopped for all active streams. The purpose of the media inactivity timer is to monitor and disconnect calls if no Real-Time Control Protocol (RTCP) packets are received within a configurable time period. However, on initiating the call hold, the originating gateway disables the media inactivity timer for that particular call, so the call remains active. The terminating gateway behaves in the same way when it receives the call-hold re-Invite from the originating gateway. When the call resumes, the originating gateway reenables the Media Inactivity Timer. For more information on the timer, refer to the SIP Media Inactivity Timer document.

All billing and accounting procedures are unaffected by the SIP: ISDN Suspend/Resume Support feature.

How to Configure Suspend and Resume Support

This section contains the following procedure:

•Configuring Suspend and Resume (required)

Configuring Suspend and Resume

Suspend and Resume functionality is enabled by default. However, Suspend and Resume functionality is also configurable. To configure Suspend and Resume for all dial peers on the VoIP network, enter the commands shown below on the originating and terminating gateways.

SUMMARY STEPS

1. enable

2. configure terminal

3. sip-ua

4. suspend-resume

DETAILED STEPS

	Command or Action	Purpose
Step 1	enable Example: Router> enable	Enables privileged EXEC mode. •Enter your password if prompted.
Step 2	configure terminal Example: Router# configure terminal	Enters global configuration mode.
Step 3	sip-ua Example: Router(config)# sip-ua	Enters SIP user agent configuration mode.
Step 4	suspend-resume Example: Router(config-sip-ua)# suspend-resume	Enables support for Suspend and Resume.

Troubleshooting Tips

To troubleshoot this feature, perform the following steps:

•Use the show sip-ua status command to display whether Suspend and Resume support is enabled or disabled.

Router# show sip-ua status

SIP User Agent Status

SIP User Agent for UDP : ENABLED

SIP User Agent for TCP : ENABLED

SIP User Agent bind status(signaling): DISABLED

SIP User Agent bind status(media): DISABLED

SIP max-forwards : 6

SIP DNS SRV version: 1 (rfc 2052)

SDP application configuration: 

 Version line (v=) required 

 Owner line (o=) required 

 Session name line (s=) required 

 Timespec line (t=) required 

 Media supported: audio image 

 Network types supported: IN 

 Address types supported: IP4 

 Transport types supported: RTP/AVP udptl

SIP support for ISDN SUSPEND/RESUME: ENABLED 

•Make sure that you can make a voice call.

•Use the debug ccsip all command to enable all SIP debugging capabilities, or use one of the following SIP debug commands:

–debug ccsip calls

–debug ccsip error

–debug ccsip events

–debug ccsip info

–debug ccsip media

–debug ccsip messages

–debug ccsip states

Configuration Examples for SIP: ISDN Suspend/Resume Support

This section provides the following configuration example:

•SIP: ISDN Suspend/Resume Support Example

SIP: ISDN Suspend/Resume Support Example

The following example contains output from the show running-config command. Because SIP Suspend and Resume is enabled by default on the gateway, this example shows SIP Suspend and Resume disabled on the gateway.

Router# show running-config

Building configuration...

Current configuration : 2791 bytes

!

version 12.2

service config

no service single-slot-reload-enable

no service pad

service timestamps debug uptime

service timestamps log uptime

no service password-encryption

service internal

service udp-small-servers

!

interface FastEthernet2/0

ip address 172.18.200.24 255.255.255.0

duplex auto

no shut

speed 10 

ip rsvp bandwidth 7500 7500

!

voice-port 1/1/1

no supervisory disconnect lcfo

!

dial-peer voice 1 pots

application session

destination-pattern 8185551111

port 1/1/1

!

dial-peer voice 3 voip

application session

destination-pattern 7175551111

session protocol sipv2

session target ipv4:172.18.200.36

codec g711ulaw

!

dial-peer voice 4 voip

application session

destination-pattern 6165551111

session protocol sipv2

session target ipv4:172.18.200.33

codec g711ulaw

!

gateway

!

sip-ua

 no suspend-resume

 retry invite 1

 retry bye 1

!

line con 0 

line aux 0 

line vty 0 4 

login 

! 

end

Additional References

For additional information related to SIP: ISDN Suspend/Resume Support, refer to the following references:

Related Documents

Related Topic	Document Title
Cisco SIP Functionality	Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2 T Session Initiation Protocol (SIP) for VoIP, Release 12.2(8)T Session Initiation Protocol Gateway Call Flows, Release 12.2(4)T Call Transfer Capabilities Using the Refer Method, Release 12.2(8)T SIP Media Inactivity Timer, Release 12.2(8)T Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events, Release 12.2(2)XB
Cisco IOS References	Cisco IOS Debug Command Reference, Release 12.2 T Cisco IOS IP Configuration Guide, Release 12.2 Cisco IOS IP Command Reference, Volume 1 of 3: Addressing and Services, Release 12.2 T Cisco IOS IP Command Reference, Volume 2 of 3: Routing Protocols, Release 12.2 T Cisco IOS IP Command Reference, Volume 3 of 3: Multicast, Release 12.2 T Cisco IOS Dial Technologies Configuration Guide, Release 12.2

Standards

Standards1	Title
ANSI T1.113	ISDN User Part
Q.764	Signalling System No. 7 - ISDN User Part signalling procedures
Q.931	ISDN user-network interface layer 3 specification for basic call control
draft-ietf-sip-isup-03.txt.	ISUP to SIP Mapping draft

1 Not all supported standards are listed.

MIBs

MIBs1	MIBs Link
•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

1 Not all supported MIBs are listed.

To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL:

http://tools.cisco.com/ITDIT/MIBS/servlet/index

If Cisco MIB Locator does not support the MIB information that you need, you can also obtain a list of supported MIBs and download MIBs from the Cisco MIBs page at the following URL:

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

To access Cisco MIB Locator, 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 found at this URL:

http://www.cisco.com/register

RFCs

RFCs1	Title
RFC 2543	SIP: Session Initiation Protocol

1 Not all supported RFCs are listed.

Technical Assistance

Description	Link
Technical Assistance Center (TAC) home page, containing 30,000 pages of searchable technical content, including links to products, technologies, solutions, technical tips, and tools. Registered Cisco.com users can log in from this page to access even more content.	http://www.cisco.com/public/support/tac/home.shtml

Command Reference

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

New Commands

•suspend-resume (SIP)

Modified Commands

•show sip-ua status

suspend-resume (SIP)

To enable SIP Suspend and Resume functionality, use the suspend-resume command in SIP user agent configuration mode. To disable SIP Suspend and Resume functionality, use the no form of this command.

suspend-resume

no suspend-resume

Syntax Description

This command has no arguments or keywords.

Defaults

Enabled

Command Modes

SIP user agent configuration

Command History

Release	Modification
12.2(15)T	This command was introduced.

Usage Guidelines

Session Initiation Protocol (SIP) gateways are now enabled to use Suspend and Resume. Suspend and Resume are basic functions of ISDN and ISDN User Part (ISUP) signaling procedures. A Suspend message temporarily halts communication (call hold), and a Resume message is received after a Suspend message and continues the communication.

Examples

The following example disables Suspend and Resume functionality:

Router(config)# sip-ua

Router(config-sip-ua)# no suspend-resume

Related Commands

Command	Description
show sip-ua status	Displays SIP UA status.
sip-ua	Enables the SIP user-agent configuration commands.

show sip-ua status

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

show sip-ua status

Syntax Description

This command has no arguments or keywords.

Command Modes

Privileged EXEC

Command History

Release	Modification
12.1(1)T	This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco AS5300.
12.1(3)T	The statistics portion of the output was removed and included in the show sip-ua statistics command.
12.2(2)XA	This command was implemented on the Cisco AS5350 and Cisco AS5400.
12.2(2)XB	Command output was enhanced to display whether media or signaling binding is enabled and the style of DNS SRV query: •1 for RFC 2052 •2 for RFC 2782
12.2(2)XB1	This command was implemented on the Cisco AS5850.
12.2(8)T	This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release. For the purposes of display, this command was separated from the generic show sip-ua command.
12.2(11)T	Command output was enhanced to display information on Session Description Protocol (SDP) application configuration. This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.
12.2(13)T	Command output was enhanced to display the following: •Information on redirection message handling. •Information on handling of 180 responses with SDP.
12.2(15)T	Command output was enhanced to display Suspend and Resume support.

Usage Guidelines

Use this command to verify SIP configurations.

Examples

The following is sample output from 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 User Agent bind status(signaling): DISABLED

SIP User Agent bind status(media): DISABLED

SIP early-media for 180 responses with SDP: DISABLED

SIP max-forwards : 6

SIP DNS SRV version: 1 (rfc 2052)

Redirection (3xx) message handling: ENABLED

SDP application configuration: 

 Version line (v=) required 

 Owner line (o=) required 

 Session name line (s=) required 

 Timespec line (t=) required 

 Media supported: audio image 

 Network types supported: IN 

 Address types supported: IP4 

 Transport types supported: RTP/AVP udptl

SIP support for ISDN SUSPEND/RESUME: ENABLED 

Table 1 describes significant fields in this output.

Table 1 show sip-ua status Field Descriptions 

Field	Description
SIP User Agent Status	UA status.
SIP User Agent for UDP	UDP is enabled or disabled.
SIP User Agent for TCP	TCP is enabled or disabled.
SIP User Agent bind status (signaling)	Binding for signaling is enabled or disabled.
SIP User Agent bind status (media)	Binding for media is enabled or disabled.
SIP early-media for 180 responses with SDP	Early media cut-through treatment for 180 responses with SDP can be enabled, the default treatment, or disabled, with local ringback provided.
SIP max-forwards	Value of max-forwards of SIP messages.
SIP DNS SRV version	Style of DNS SRV query: 1 for RFC 2052 or 2 for RFC 2782.
Redirection (3xx) message handling	Redirection can be enabled, which is the default status as per RFC 2543. Or handling of redirection 3xx messages can be disabled, allowing the gateway to treat 3xx redirect messages as 4xx error messages.
Version	SDP version.
Owner	Session originator.
Session name	Identifies the session name.
Timespec	Identifies the session start and stop times.
Media supported	Media information.
Network types supported	Always IN for Internet.
Address types supported	Identifies the Internet Protocol version.
Transport type supported	Identifies the transport protocols supported.
SIP support for ISDN SUSPEND/RESUME	Suspend and Resume support is enabled or disabled.

Related Commands

Command	Description
show sip-ua retry	Displays SIP retry statistics.
show sip-ua statistics	Displays response, traffic, and retry SIP statistics.
show sip-ua timers	Displays the current settings for SIP UA timers.
sip-ua	Enables the SIP user-agent configuration commands.

Glossary

H.323—An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.

ISDN—Integrated Services Digital Network. Communication protocol offered by telephone companies that permits telephone networks to carry data, voice, and other source traffic.

ISUP—ISDN User Part. Provides the interexchange signaling to support SS7 trunks that are set up for switched voice and data applications in an ISDN environment.

OGW—Cisco originating gateway.

PSTN—public switched telephone network.

RES—ISUP Resume message defined in ITU Q.764.

Resume—ISDN Resume message defined in ITU Q.931.

RTCP—RTP Control Protocol. The protocol monitors an RTP connection and conveys information about the ongoing session.

SDP—Session Description Protocol. Messages containing capabilities information that are exchanged between gateways.

SIP—Session Initiation Protocol. An application-layer protocol originally developed by the Multiparty Multimedia Session Control (MMUSIC) working group of the Internet Engineering Task Force (IETF). Their goal was to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.

SUS—ISUP Suspend message defined in ITU Q.764.

Suspend—ISDN Suspend message defined in ITU Q.931.

TGW—Cisco terminating gateway.

UA—user agent. A combination of UAS and UAC that initiates and receives calls. See also UAS and UAC.

UAC—user agent client. A client application that initiates a SIP request.

UAS—user agent server. A server application that contacts the user when a SIP request is received and then returns a response on behalf of the user. The response accepts, rejects, or redirects the request.

VoIP—Voice over IP. The ability to carry normal telephone-style voice over an IP-based network.

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Note Refer to the Internetworking Terms and Acronyms for terms not included in this glossary.

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