H.323-to-SIP Connections on a Cisco Unified Border Element

First Published: June 19, 2006
Last Updated: November 17, 2010

This chapter describes how to configure and enable features for H.323-to-SIP connections in a Cisco Unified Border Element topology.

Activation Cisco Product Authorization Key (PAK)—A Product Authorization Key (PAK) is required to configure some of the features described in this guide. Before you start the configuration process, please register your products and activate your PAK at the following URL http://www.cisco.com/go/license.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the "Feature Information for H.323-to-SIP Connections on a Cisco Unified Border Element" section.

Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Contents

•Prerequisites for Configuring H.323-to-SIP Connection on a Cisco Unified Border Element

•Restrictions for Configuring H.323-to-SIP Connections on a Cisco Unified Border Element

•Information About H.323-to-SIP Connections on a Cisco Unified Border Element

•How to Configure H.323-to-SIP Connections on a Cisco Unified Border Element

•Where to Go Next

•Additional References

•Feature Information for H.323-to-SIP Connections on a Cisco Unified Border Element

Prerequisites for Configuring H.323-to-SIP Connection on a Cisco Unified Border Element

•Perform the prerequisites listed in the "Prerequisites for Cisco Unified Border Element Configuration" section in this guide.

•Perform fundamental gateway configuration listed in the "Prerequisites for Fundamental Cisco Unified Border Element Configuration" section in this guide.

•Perform basic H.323 gateway configuration.

•Perform basic H.323 gatekeeper configuration.

Note For configuration instructions, see the "Configuring H.323 Gateways" and "Configuring H.323 Gatekeepers" chapters of the Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2.

Restrictions for Configuring H.323-to-SIP Connections on a Cisco Unified Border Element

•Changing codecs during rotary dial peer selection is not supported.

•Codec preference order in voice class should be the same in all dial peers.

•Configure extended capabilities on dial peers for fast start-to-early media scenarios.

•Delayed Offer to Slow-Start is not supported for SRTP-to-SRTP H.323-to-SIP calls.

•During a triggered INVITE scenario the Cisco UBE always generates a delayed offer INVITE.

•Fast-start to delayed-media signal interworking is not supported.

•Fast Start to Early Offer Supplementary Service will not work without extended capabilities configured under dial-peer.

•GSMFR and GSMEFR codecs are not supported.

•H450.2 & H450.3 are enabled & invisible under dial peers by default. H.450 cannot be enabled at the dial peer level if they are globally disabled.

•Media flow-around is not supported.

•Passing multiple diversion headers or multiple contact header in 302 to the H.323 leg is not supported.

•RSVP for supplementary scenarios is not supported.

•Session refresh is not supported.

•SIP-to-H.323 Supplementary Services based on H.450 is not supported.

•Slow-start to early media signal interworking is not supported.

•Supplementary services are Empty Capability Set (ECS) based supplementary services from the H.323 perspective, not H.450 supplementary services.

•Transcoding for supplementary calls is not supported.

•DTMF Interworking rtp-nte to out of band is not supported when high density transcoder is enabled. Use normal transcoding for rtp-nte to out of band DTMF interworking.

Cisco IOS Release 12.4(15)XY and earlier releases:

•SRTP Passthrough is not supported.

Cisco IOS Release 12.4(11)XJ2 and earlier releases:

•Delayed-media to slow-start signal interworking is not supported.

•H323-SIP Supplementary Services is not supported (ECS based).

Cisco IOS Release 12.4(11)T and earlier releases:

•Codec Transparent is not supported.

Cisco IOS Release 12.4(2)T and earlier releases:

•Extended codec support and codec filtering is not supported.

Cisco IOS Release 12.3(8)T and earlier releases:

•Basic call is not supported.

Information About H.323-to-SIP Connections on a Cisco Unified Border Element

•All codecs using static payload are supported.

•Fast-start to early media signal interworking is supported.

•H.323-to-SIP Supplementary Services are supported in Cisco IOS Release12.4(15)XY and later.

•Supported codecs using dynamic payload are g726r16 and g726r24.

•Slow-start to delayed-media signal interworking is supported.

•One or multiple codes may configured on the incoming and out-going dial-peer.

•SRTP-to-SRTP for SIP-to-H.323 calls is supported:

–Supported signal interworking include: Fast-Start to Early Offer, Early Offer to Fast-Start, and Slow-Start to Delayed Offer.

How to Configure H.323-to-SIP Connections on a Cisco Unified Border Element

The section contains the following tasks:

•H.323-to-SIP Basic Call Interworking for Session Border Controller (SBC)

•H.323-to-SIP Supplementary Feature Interworking for Session Border Controller (SBC)

•H.323-to-SIP Supplementary Service Enhancements for Session Border Controller (SBC)

•Configuring H.323-to-SIP Connections on a Cisco Unified Border Element

•Configuring DTMF Relay Digit-Drop on a Cisco Unified Border Element

•Configuring H.323-to-SIP Call Failure Recovery (Rotary) on a Cisco Unified Border Element

•Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks

•Managing H.323 IP Group Call Capacities

•Troubleshooting and Verifying H.323-to-SIP connections on a Cisco Unified Border Element

H.323-to-SIP Basic Call Interworking for Session Border Controller (SBC)

This feature enables the IP-to-IP gateway to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323). The SIP-to-H.323 protocol interworking capabilities of the Cisco Unified Border Element support the following:

•Basic voice calls (G.711 and G.729 codecs)

•UDP and TCP transport

•Interworking between H.323 Fast-Start and SIP early-media signaling

•Interworking between H.323 Slow-Start and SIP delayed-media signaling

•DTMF relay interworking:

–H.245 alpha/signal <--> SIP RFC 2833

–H.245 alpha/signal <--> SIP Notify

–H.245 alpha/signal <--> SIP KPML

•Codec transcoding (G.711-G.729)

•Calling/called name and number

•T.38 fax relay and Cisco fax relay

•RADIUS call-accounting records

•RSVP synchronized with call signaling

•TCL IVR 2.0 for SIP, including media playout and digit collection (RFC 2833 DTMF relay)

H.323-to-SIP Supplementary Feature Interworking for Session Border Controller (SBC)

Provides enhanced termination and re-origination of signaling and media between VoIP and Video Networks in conformance with RFC3261. New features offered in this release on the Cisco 28xx, 38xx, 5350XM and 5400XM include:

•Support H.323-to-SIP Supplementary services for Cisco Unified Communications Manager with MTP on the H.323 Trunk.

•ILBC Codec Support

•Interworking between G.711 inband DTMF to RFC2833

•VXML 3.x support

•VXML support with SIP Notify

Restrictions

•H450.2 & H450.3 are enabled & invisible under dial peers by default. H.450 cannot be enabled at the dial peer level if they are globally disabled.

•H.245 signal/alpha <--> SIP Raw In band is not supported.

•RSVP for supplementary scenarios is not supported.

•Transcoding for supplementary calls is not supported.

H.323-to-SIP Supplementary Service Enhancements for Session Border Controller (SBC)

H.323-to-SIP features offered in this release include:

•Mapping ECS to ReINVITE and ECS to REFER on the Cisco IOS SBC.

Configuring H.323-to-SIP Connections on a Cisco Unified Border Element

To configure H.323-to-SIP connections on a Cisco Unified Border Element. perform the steps in this section.

Restrictions

Connections are disabled by default in Cisco IOS images that support the Cisco Unified Border Element

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. allow-connections

5. exit

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 

voice service voip

Example:

Router(config)# voice service voip

Enters VoIP voice-service configuration mode.

Step 4 

allow-connections from-type to to-type

Example:

Router(conf-voi-serv)# allow-connections h323 to sip

Allows connections between specific types of endpoints in an Cisco Unified Border Element. Arguments are as follows:

•from-type—Type of connection. Valid values: h323, sip.

•to-type—Type of connection. Valid values: h323, sip.

Note H.323-to-H.323: By default, H.323-to-H.323 connections are disabled and POTS-to-any and any-to-POTS connections are enabled.

Step 5 

exit

Example:

Router(conf-voi-serv)# exit

Exits the current mode.

Configuring DTMF Relay Digit-Drop on a Cisco Unified Border Element

To avoid sending both in-band and out-of band tones to the outgoing leg when sending Cisco Unified Border Element calls in-band (rtp-nte) to out-of band (h245-alphanumeric). Configure the dtmf-relay rtp-nte digit-drop command on the incoming SIP dial-peer. On the H.323 side configure either dtmf-relay h245-alphanumeric or dtmf-relay h245-signal. This may also be used for H.323-to-SIP calls.

To configure DTMF relay digit drop on an Cisco Unified Border Element, perform the steps in this section.

Restrictions

•The debug output will show that the H245 out of band messages are sent to the TGW. However, the digits are not heard on the phone.

•Cisco UBE same dial-peer for match incoming-called from H.323 or SIP is not supported.

For additional information on DTMF relay capabilities. See the "Configuring DTMF Relay and Payload Type" section of the Dial Peer Configuration on Voice Gateway Routers Configuration Guide

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. dtmf-relay [cisco-rtp] [h245-alphanumeric] [rtp-nte [digit-drop]]

5. exit

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 

dial-peer voice number voip

Example:

Router(config)# dial-peer voice 2 voip

Enters dial-peer configuration mode for the specified VoIP dial peer.

Step 4 

dtmf-relay [cisco-rtp] [h245-alphanumeric] [rtp-nte [digit-drop]]

Example:

Router (config-dial-peer)# dtmf-relay rtp-nte digit-drop h245-alphanumeric

Forwards DTMF tones. Keywords are as follows:

•cisco-rtp—Forwards DTMF tones by using RTP with a Cisco-proprietary payload type.

•h245-alphanumeric—Forwards DTMF tones by using the H.245 alphanumeric method.

•h245-signal—Forwards DTMF tones by using the H.245 signal UII method.

•rtp-nte—Forwards DTMF tones by using Real-Time Transport Protocol (RTP) with the Named Telephone Event (NTE) payload type.

•digit-drop—Passes digits out-of-band, and in-band digits are dropped.

Note The digit-drop keyword is only seen went the rtp-nte keyword is configured.

Step 5 

exit

Example:

Router(config-dial-peer)# exit

Exits the current mode.

Examples

The following example shows DTMF-Relay digits configured to avoid sending both in-band and out-of-band tones to the outgoing leg in an Cisco Unified Border Element: 

.

.

.

dial-peer voice 1 voip

 voice-class codec 2

 dtmf-relay rtp-nte digit-drop h245-alphanumeric

.

.

.

Configuring H.323-to-SIP Call Failure Recovery (Rotary) on a Cisco Unified Border Element

Call failure recovery (Rotary) on the Cisco Unified Border Element eliminates the need for identical codec capabilities for all dial peers in the rotary group, and allows the Cisco Unified Border Element to restart the codec negotiation end-to-end. Call failure recovery will continue until "voice hunt stop" is reached.

To configure H.323-to-SIP call failure recovery (rotary) on an Cisco Unified Border Element, perform the steps in this section.

Restrictions

If extended caps (DTMF or T.38) are configured on the outgoing gateway or the trunking gateway, extended caps must be configured in both places.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. h323

5. emptycapability

6. exit

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 

voice service voip

Example:

Router(config)# voice service voip

Enters VoIP voice-service configuration mode.

Step 4 

h323

Example:

Router(conf-voi-serv)# h323

Enters H.323 voice-service configuration mode.

Step 5 

emptycapability

Example:

Router(conf-serv-h323)# emptycapability

Enables call failure recovery (TCS=0).

Step 6 

exit

Example:

Router(conf-serv-h323)# exit

Exits the current mode.

Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks

The Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature provides precondition-based Resource Reservation Protocol (RSVP) support for basic audio call and supplementary services on Cisco UBE. This feature improves the interoperability between RSVP and non-RSVP networks. RSVP functionality added to Cisco UBE helps you to reserve the required bandwidth before making a call.

This feature extends RSVP support to delayed-offer to delayed-offer and delayed-offer to early-offer calls, along with the early-offer to early-offer calls.

Prerequisites

To enable this feature, you must have Cisco IOS Release 15.0(1)XA or a later release installed and running on your Cisco gateway. For detailed information on platform availability and subsequent releases, see the "Feature Information for H.323-to-SIP Connections on a Cisco Unified Border Element" section.

RSVP policies allow you to configure separate bandwidth pools with varying limits so that any one application, such as video, can consume all the RSVP bandwidth on a specified interface at the expense of other applications, such as voice, which would be dropped.

To limit bandwidth per application, you must configure a bandwidth limit before configuring Support for the Interworking Between RSVP Capable and RSVP Incapable Networks feature. See the "Configuring RSVP on an Interface" section.

Restrictions

The Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature has the following restrictions:

•Segmented RSVP is not supported.

•Interoperability between Cisco UBE and Cisco Unified Communications Manager is not available.

•RSVP-enabled video calls are not supported.

Configuring RSVP on an Interface

You must allocate some bandwidth for the interface before enabling RSVP. Perform this task to configure RSVP on an interface.

SUMMARY STEPS

1. enable

2. configure terminal

3. interface type slot/port

4. ip rsvp bandwidth [reservable-bw [max-reservable-bw] [sub-pool reservable-bw]]

5. end

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 

interface type slot/port

Example:

Router(config)# interface FastEthernet 0/1

Configures an interface type and enters interface configuration mode.

Step 4 

ip rsvp bandwidth [reservable-bw [max-reservable-bw] [sub-pool reservable-bw]]

Example:

Router(config-if)# ip rsvp bandwidth 10000 100000

Enables RSVP for IP on an interface.

Step 5 

end

Example:

Router(config-if)# end

(Optional) Exits interface configuration mode and returns to privileged EXEC mode.

Configuring Optional RSVP on the Dial Peer

Perform this task to configure optional RSVP at the dial peer level. This configuration allows you to have uninterrupted call even if there is a failure in bandwidth reservation.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. no acc-qos {controlled-load | guaranteed-delay} [audio | video]

5. req-qos {controlled-load | guaranteed-delay} [audio | video] [bandwidth [default bandwidth-value] [max bandwidth-value]]

6. end

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer 77 voip

Enters dial peer voice configuration mode.

Step 4 

no acc-qos {controlled-load | guaranteed-delay} [audio | video]

Example:

Router(config-dial-peer)# no acc-qos controlled-load

Removes any value configured for the acc-qos command.

–controlled-load—Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded.

–guaranteed-delay—Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.

Step 5 

req-qos {controlled-load | guaranteed-delay} [audio | video] [bandwidth [default bandwidth-value] [max bandwidth-value]]

Example:

Router(config-dial-peer)# req-qos controlled-load

Configures the desired quality of service (QoS) to be used.

•Calls continue even if there is a failure in bandwidth reservation.

Note Configure the req-qos command using the same keyword that you used to configure the acc-qos command, either controlled-load or guaranteed-delay. That is, if you configured acc-qos controlled-load command in the previous step, then use the req-qos controlled-load command here.

Step 6 

end

Example:

Router(config-dial-peer)# end

Exits dial peer voice configuration mode and returns to privileged EXEC mode.

Configuring EO to EO, DO to DO and DO to EO at the Dial Peer

Perform this task to configure support for EO to EO, DO to DO, and DO to EO at the dial peer level.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. no acc-qos {controlled-load | guaranteed-delay} [audio | video]

5. req-qos {controlled-load | guaranteed-delay} [audio | video] [bandwidth [default bandwidth-value] [max bandwidth-value]]

6. exit

7. interface type slot/port

8. ip rsvp bandwidth [reservable-bw [max-reservable-bw] [sub-pool reservable-bw]]

9. exit

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 77 voip

Enters dial peer voice configuration mode.

Step 4 

no acc-qos {controlled-load | guaranteed-delay} [audio | video]

Example:

Router(config-dial-peer)# no acc-qos controlled-load

Removes any value configured for the acc-qos command.

–controlled-load—Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded.

–guaranteed-delay—Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.

Step 5 

req-qos {controlled-load | guaranteed-delay} [audio | video] [bandwidth [default bandwidth-value] [max bandwidth-value]]

Example:

Router(config-dial-peer)# req-qos controlled-load

Configures the desired quality of service (QoS) to be used.

•Calls continue even if there is a failure in bandwidth reservation.

Note Configure the req-qos command using the same keyword that you used to configure the acc-qos command, either controlled-load or guaranteed-delay. That is, if you configured the acc-qos controlled-load command in the previous step, then use the req-qos controlled-load command here.

Step 6 

exit

Example:

Router(config-dial-peer)# exit

Exits dial peer voice configuration mode and returns to global configuration mode.

Step 7 

interface type slot/port

Example:

Router(config)# interface FastEthernet 0/1

Configures an interface type and enters interface configuration mode.

Step 8 

ip rsvp bandwidth [reservable-bw [max-reservable-bw] [sub-pool reservable-bw]]

Example:

Router(config-if)# ip rsvp bandwidth 10000 100000

Enables RSVP for IP on an interface.

Step 9 

exit

Example:

Router(config-if)# exit

Exits interface configuration mode and returns to privileged EXEC mode.

Configuring Mandatory RSVP on the Dial Peer

Perform this task to configure Mandatory RSVP on the dial peer. This configuration ensures that the call does not connect if sufficient bandwidth is not allocated.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. acc-qos {best-effort | controlled-load | guaranteed-delay} [audio | video]

5. req-qos {best-effort [audio | video] | {controlled-load | guaranteed-delay} [audio | video] [bandwidth [default bandwidth-value] [max bandwidth-value]]}

6. end

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer 77 voip

Enters dial peer voice configuration mode.

Step 4 

acc-qos {best-effort | controlled-load | guaranteed-delay} [audio | video]

Example:

Router(config-dial-peer)# acc-qos best-effort

Configures mandatory RSVP on the dial-peer.

–best-effort—Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. This is the default.

–controlled-load—Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded.

–guaranteed-delay—Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queueing if the bandwidth reserved is not exceeded.

Step 5 

req-qos {best-effort [audio | video] | {controlled-load | guaranteed-delay} [audio | video] [bandwidth [default bandwidth-value] [max bandwidth-value]]}

Example:

Router(config-dial-peer)# req-qos controlled-load

Configures mandatory RSVP on the dial-peer.

•Calls continue even if there is a drop in the bandwidth reservation.

Step 6 

end

Example:

Router(config-dial-peer)# end

(Optional) Exits dial peer voice configuration mode and returns to privileged EXEC mode.

Configuring Midcall RSVP Failure Policies

Perform this task to enable call handling policies for a midcall RSVP failure.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. voice-class sip rsvp-fail-policy {video | voice} post-alert {optional keep-alive | mandatory {keep-alive | disconnect retry retry-attempts}} interval seconds

5. end

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 66 voip

Enters dial peer voice configuration mode.

Step 4 

voice-class sip rsvp-fail-policy {video | voice} post-alert {optional keep-alive | mandatory {keep-alive | disconnect retry retry-attempts}} interval seconds

Example:

Router(config-dial-peer)# voice-class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 50

Enables call handling policies for a midcall RSVP failure.

–optional keep-alive—The keepalive messages are sent when RSVP fails only if RSVP negotiation is optional.

–mandatory keep-alive—The keepalive messages are sent when RSVP fails only if RSVP negotiation is mandatory.

Note Keepalive messages are sent at 30-second intervals when a postalert call fails to negotiate RSVP regardless of the RSVP negotiation setting (mandatory or optional).

Step 5 

end

Example:

Router(config-dial-peer)# end

Exits dial peer voice configuration mode and returns to privileged EXEC mode.

Configuring DSCP Values

Perform this task to configure different Differentiated Services Code Point (DSCP) values based on RSVP status.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. ip qos dscp {dscp-value | set-af | set-cs | default | ef} {signaling | media [rsvp-pass | rsvp-fail] | video [rsvp-none | rsvp-pass | rsvp-fail]}

5. end

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 66 voip

Enters dial peer voice configuration mode.

Step 4 

ip qos dscp {dscp-value | set-af | set-cs | default | ef} {signaling | media [rsvp-pass | rsvp-fail] | video [rsvp-none | rsvp-pass | rsvp-fail]}

Example:

Router(config-dial-peer)# ip qos dscp af11 media rsvp-pass

Configures DSCP values based on RSVP status.

–media rsvp-pass—Specifies that the DSCP value applies to media packets with successful RSVP reservations.

–media rsvp-fail—Specifies that the DSCP value applies to packets (media or video) with failed RSVP reservations.

–The default DSCP value for all media (voice and fax) packets is ef.

Note You must configure the DSCP values for all cases: media rsvp-pass and media rsvp-fail.

Step 5 

end

Example:

Router(config-dial-peer)# end

Exits dial peer voice configuration mode and returns to privileged EXEC mode.

Configuring an Application ID

Perform this task to configure a specific application ID for RSVP establishment.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. ip qos policy-locator {video | voice} [app app-string] [guid guid-string] [sapp subapp-string] [ver version-string]

5. end

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 66 voip

Enters dial peer voice configuration mode.

Step 4 

ip qos policy-locator {video | voice} [app app-string] [guid guid-string] [sapp subapp-string] [ver version-string]

Example:

Router(config-dial-peer)# ip qos policy-locator voice

Configures a QoS policy locator (application ID) used to deploy RSVP policies for specifying bandwidth reservations on Cisco IOS Session Initiation Protocol (SIP) devices.

Step 5 

end

Example:

Router(config-dial-peer)# end

Exits dial peer voice configuration mode and returns to privileged EXEC mode.

Configuring Priority

Perform this task to configure priorities for call preemption.

SUMMARY STEPS

1. enable

2. configure terminal

3. dial-peer voice tag voip

4. ip qos defending-priority defending-pri-value

5. ip qos preemption-priority preemption-pri-value

6. end

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

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 

dial-peer voice tag voip

Example:

Router(config)# dial-peer voice 66 voip

Enters dial peer voice configuration mode.

Step 4 

ip qos defending-priority defending-pri-value

Example:

Router(config-dial-peer)# ip qos defending-priority 66

Configures the RSVP defending priority value for determining QoS.

Step 5 

ip qos preemption-priority preemption-pri-value

Example:

Router(config-dial-peer)# ip qos preemption-priority 75

Configures the RSVP preemption priority value for determining QoS.

Step 6 

end

Example:

Router(config-dial-peer)# end

Exits dial peer configuration mode and returns to privileged EXEC mode.

Troubleshooting the Support for Interworking Between RSVP Capable and RSVP Incapable Networks Feature

Use the following commands to debug any errors that you may encounter when you configure the Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature.

•debug call rsvp-sync events

•debug call rsvp-sync func-trace

•debug ccsip all

•debug ccsip messages

•debug ip rsvp messages

•debug sccp all

Verifying Support for Interworking Between RSVP Capable and RSVP Incapable Networks

This task explains how to display information to verify the configuration for the Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature. These commands need not be entered in any specific order.

SUMMARY STEPS

1. enable

2. show sip-ua calls

3. show ip rsvp installed

4. show ip rsvp reservation

5. show ip rsvp interface detail [interface-type number]

6. show sccp connections details

7. show sccp connections rsvp

8. show sccp connections internal

9. show sccp [all | connections | statistics]

DETAILED STEPS
 
Command or Action
Purpose

Step 1 

enable

Example:

Router> enable

Enables privileged EXEC mode.

•Enter your password if prompted.

Step 2 

show sip-ua calls

Example:

Router# show sip-ua calls

(Optional) Displays active user agent client (UAC) and user agent server (UAS) information on SIP calls.

Step 3 

show ip rsvp installed

Example:

Router# show ip rsvp installed

(Optional) Displays RSVP-related installed filters and corresponding bandwidth information.

Step 4 

show ip rsvp reservation

Example:

Router# show ip rsvp reservation

(Optional) Displays RSVP-related receiver information currently in the database.

Step 5 

show ip rsvp interface detail [interface-type number]

Example:

Router# show ip rsvp interface detail GigabitEthernet 0/0

(Optional) Displays the interface configuration for hello.

Step 6 

show sccp connections details

Example:

Router# show sccp connections details

(Optional) Displays SCCP connection details, such as call-leg details.

Step 7 

show sccp connections rsvp

Example:

Router# show sccp connections rsvp

(Optional) Displays information about active SCCP connections that are using RSVP.

Step 8 

show sccp connections internal

Example:

Router# show sccp connections internal

(Optional) Displays the internal SCCP details, such as time-stamp values.

Step 9 

show sccp [all | connections | statistics]

Example:

Router# show sccp statistics

(Optional) Displays SCCP information, such as administrative and operational status.

Managing H.323 IP Group Call Capacities

The Cisco Unified Border Element feature works with the voice source-group command to provide matching criteria for incoming calls. The voice source-group command assigns a name to a set of source IP group characteristics. The terminating gateway uses these characteristics to identify and translate the incoming VoIP call. If there is no voice source group match, the default carrier ID is used, any source carrier ID on the incoming message is transmitted without change, and no destination carrier is available. Call-capacity information is reported to the gatekeeper, but carrier routing information is not.

If the voice source group matches, the matched source carrier ID is used and the target carrier ID defined in the voice source group is used for the destination carrier ID.

To configure H.323 IP call capabilities, perform the steps in this section.

Restrictions

You can use the commands that follow only when no calls are active. If you try to use these commands with active calls present, the commands are rejected.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip

4. h323

5. ip circuit max-calls

6. ip circuit carrier-id

7. ip circuit default only

8. ip circuit default name

9. exit

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 

voice service voip

Example: 
Router(config)# voice service voip

Enters VoIP voice-service configuration mode.

Step 4 

h323

Example:

Router(config-voice-service)# h323

Enters H.323 voice-service configuration mode.

Step 5 

ip circuit carrier-id carrier-name [reserved-calls reserved]

Example:

Router(config-serv-h323)# ip circuit carrier-id AA reserved-calls 500

(Optional) Defines an IP circuit using the specified name as the circuit ID.

Note The reserved keyword for this command is optional. Using this keyword creates a specified maximum number of calls for that circuit ID. The default value is 200 call legs.

Step 6 

ip circuit default only

Example:

Router(config-serv-h323)# ip circuit default only

(Optional) Creates a single carrier to use all of the call capacity available to the Cisco Unified Border Element.

Note If you use the ip circuit default only command, you cannot use the ip circuit carrier-id command to configure more circuits. Using the ip circuit default only command creates a single carrier using the default carrier name.

Step 7 

ip circuit default name carrier-name

Example:

Router(config-serv-h323)# ip circuit default name AA

(Optional) Changes the default circuit name.

Step 8 

exit

Example:

Router(config-serv-h323)# exit

Exits the current mode.

Examples

The following examples show a default carrier with no voice source group configured:

Default Carrier with No Voice Source Group 

voice service voip

 allow-connections h323 to h323

 h323

  ip circuit max-calls 1000

  ip circuit default only

 
 
 
  
 
 
  
 
 
 
  
 
 If there is no incoming source carrier ID: 
 
  
 
 •Capacity only is reported to the gatekeeper using the default circuit (two call legs). 
 
  
 
 •No source or destination carrier information is reported. 
 
  
 
 If there is an incoming source carrier ID: 
 
  
 
 •Two call legs are counted against the default circuit and reported to the GK. 
 
  
 
 •The source carrier ID is passed through the gateway to the terminating leg. 
 
  
 
 The following examples show a configuration with more reserved calls than the default value for the max-calls argument (1000): 
 
  
  
 
 Configuration with Default Calls in Excess of 1000 
 
  
 
 This example assigns 1100 calls to other carriers, leaving 400 calls available to the default carrier: 
 
  
 
 
  
voice service voip

 
 
 
  
 
 
  
 allow-connections h323 to h323

 
 
 
  
 
 
  
 h323

 
 
 
  
 
 
  
  ip circuit max-calls 1000

 
 
 
  
 
 
  
  ip circuit carrier-id AA reserved-calls 500

 
 
 
  
 
 
  
  ip circuit carrier-id bb reserved-calls 500

 
 
 
  
 
 
  
  ip circuit carrier-id cc reserved-calls 100

 
 
 
  
 
 
  
 
 
 
  
 
 The following examples show the default carrier configured with an incoming source carrier but no voice source group configured. 
 
 
 
   
 
 
 
 
  
 
 Note In this example, 800 call legs are implicitly reserved for the default circuit. 
 
 
 
  
  
 
 Default Carrier and Incoming Source Carrier with No Voice Source Group 
 
 
 
   
 
 
 
 
  
 
 Note A gatekeeper is required with carrier-id routing. 
 
 
 
  
 
 
  
voice service voip

 
 
 
  
 
 
  
 allow-connections h323 to h323

 
 
 
  
 
 
  
 h323

 
 
 
  
 
 
  
  ip circuit max-calls 1000

 
 
 
  
 
 
  
  ip circuit carrier-id AA reserved-calls 200

 
 
 
  
 
 
  
 
 
 
  
 
 If there is no incoming source carrier ID: 
 
  
 
 •Capacity only is reported to the GK using the default circuit (two call legs). 
 
  
 
 •No source or destination carrier information is reported. 
 
  
 
 If there is an incoming source carrier ID called "AA": 
 
  
 
 •One call leg is counted against circuit "AA". 
 
  
 
 •One call leg (outbound) is counted against the default circuit. 
 
  
 
 •The source carrier ID is passed through the gateway to the terminating leg. 
 
  
 
 If there is an incoming source carrier ID called "BB" (for example) or anything other than "AA": 
 
  
 
 •Two call legs are counted against the default circuit. 
 
  
 
 •The source carrier ID "BB" is passed through the gateway to the terminating leg. 
 
  
 
 The following examples show the first voice source-group match case: 
 
  
  
 
 Voice Source-Group Match Case 1 
 
  
 
 
  
voice service voip

 
 
 
  
 
 
  
 allow-connections h323 to h323

 
 
 
  
 
 
  
 h323

 
 
 
  
 
 
  
  ip circuit max-calls 1000

 
 
 
  
 
 
  
  ip circuit carrier-id AA reserved-calls 200

 
 
 
  
 
 
  
!

 
 
 
  
 
 
  
voice source-group 1

 
 
 
  
 
 
  
 carrier-id source AA

 
 
 
  
 
 
  
 carrier-id target AA

 
 
 
  
 
 
  
 
 
 
  
 
 If there is no incoming source carrier ID, the default circuit is used because there is no match in the voice source group. 
 
  
 
 If there is an incoming source carrier ID called "AA," the following are in effect: 
 
  
 
 •The voice source group matches. 
 
  
 
 •Both call legs are counted against circuit "AA". 
 
  
 
 •The source carrier ID is passed through the gateway to the terminating leg. 
 
  
 
 •The destination carrier ID is "AA". 
 
  
 
 The following examples show the second voice source group match case: 
 
  
  
 
 Voice Source-Group Match Case  2 
 
  
 
 
  
voice service voip

 
 
 
  
 
 
  
 allow-connections h323 to h323

 
 
 
  
 
 
  
 h323

 
 
 
  
 
 
  
  ip circuit max-calls 1000

 
 
 
  
 
 
  
  ip circuit carrier-id AA reserved-calls 200

 
 
 
  
 
 
  
  ip circuit carrier-id BB reserved-calls 200

 
 
 
  
 
 
  
!

 
 
 
  
 
 
  
voice source-group 1

 
 
 
  
 
 
  
 carrier-id source AA

 
 
 
  
 
 
  
 carrier-id target BB

 
 
 
  
 
 
  
 
 
 
  
 
 If there is no incoming source carrier ID, the default circuit is used because there is no match in the voice source group. 
 
  
 
 If there is an incoming source carrier ID called "AA": 
 
  
 
 •The voice source-group matches. 
 
  
 
 •One leg is counted against circuit "AA". 
 
  
 
 •One leg is counted against circuit "BB". 
 
  
 
 •The source carrier ID is passed through the gateway to the terminating leg. 
 
  
 
 •The destination carrier ID is "BB". 
 
  
 
 The following examples show the third voice source-group match case: 
 
  
  
 
 Voice Source-Group Match Case 3 
 
  
 
 
  
voice service voip

 
 
 
  
 
 
  
 allow-connections h323 to h323

 
 
 
  
 
 
  
 h323

 
 
 
  
 
 
  
  ip circuit max-calls 1000

 
 
 
  
 
 
  
  ip circuit carrier-id AA reserved-calls 200

 
 
 
  
 
 
  
  ip circuit carrier-id BB reserved-calls 200

 
 
 
  
 
 
  
!

 
 
 
  
 
 
  
voice source-group 1

 
 
 
  
 
 
  
 access-list 1

 
 
 
  
 
 
  
 carrier-id source BB

 
 
 
  
 
 
  
 
 
 
  
 
 If the access-list matches, the following apply: 
 
  
 
 •One leg is counted against circuit "BB". 
 
  
 
 •One leg is counted against the default circuit (for the destination circuit). 
 
  
 
 •The source carrier ID is synthesized to "BB" and used to report to the gatekeeper. It is also used on the outgoing setup. 
 
  
 
 •If a source carrier ID is received on the incoming setup, it is overridden with the synthesized carrier ID. 
 
   
  
  
 
 Troubleshooting and Verifying H.323-to-SIP connections on a Cisco Unified Border Element 
 
  
 
 To troubleshoot or verify connections in an Cisco Unified Border Element, perform the steps in this section. This section contains the following subsections: 
 
  
 
 •Troubleshooting Tips 
 
  
 
 •Verifying Cisco Unified Border Element Configuration and Operation 
 
   
  
  
 
 Troubleshooting Tips 
 
 
 
   
 
 
 
 
  
 
 
  Caution  
  Under moderate traffic loads, these 
  debug commands produce a high volume of output. 
 
 
 
 
  
 
 •Use the debug voip ipipgw command to debug the Cisco Unified Border Element feature. 
 
  
 
 •Use any of the following additional debug commands on the gateway as appropriate: 
 
  
 
 –debug cch323 all  
 
  
 
 –debug ccsip all  
 
  
 
 –debug h225 asn1  
 
  
 
 –debug h225 events  
 
  
 
 –debug h245 asn1  
 
  
 
 –debug h245 events  
 
  
 
 –debug voip ipipgw  
 
  
 
 –debug voip ccapi inout  
 
 
 
   
 
 
 
 
  
 
 Note For examples of show and debug command output and details on interpreting the output, see the following resources: 
 
  
 
 •Cisco IOS Debug Command Reference, Release 12.4T 
 
  
 
 •Cisco IOS Voice Troubleshooting and Monitoring Guide 
 
  
 
 •Troubleshooting and Debugging VoIP Call Basics 
 
  
 
 •VoIP Debug Commands 
 
 
 
  
   
  
  
 
 Verifying Cisco Unified Border Element Configuration and Operation 
 
  
 
 To verify Cisco Unified Border Element IP-to-IP feature configuration and operation, perform the following steps (listed alphabetically) as appropriate. 
 
 
 
   
 
 
 
 
  
 
 Note The word "calls" refers to call legs in some commands and output. 
 
 
 
   
  
 
 SUMMARY STEPS 
 
   
  
 
 DETAILED STEPS 
 
 
 
  
 
 Step 1 show call active video 
 
  
 
 Use this command to display the active video H.323 call legs. 
 
  
 
 Step 2 show call active voice 
 
  
 
 Use this command to display call information for voice calls that are in progress. 
 
  
 
 Step 3 show call active fax 
 
  
 
 Use this command to display the fax transmissions that are in progress. 
 
  
 
 Step 4 show call history video 
 
  
 
 Use this command to display the history of video H.323 call legs. 
 
  
 
 Step 5 show call history voice 
 
  
 
 Use this command to display the history of voice call legs. 
 
  
 
 Step 6 show call history fax 
 
  
 
 Use this command to display the call history table for fax transmissions that are in progress. 
 
  
 
 Step 7 show crm 
 
  
 
 Use this command to display the carrier ID list or IP circuit utilization. 
 
  
 
 Step 8 show dial-peer voice 
 
  
 
 Use this command to display information about voice dial peers. 
 
  
 
 Step 9 show running-config 
 
  
 
 Use this command to verify which H.323-to-H.323, H.323-to-SIP, or SIP-to-SIP connection types are supported. 
 
  
 
 Step 10 show voip rtp connections 
 
  
 
 Use this command to display active Real-Time Transport Protocol (RTP) connections. 
 
 
 
  
   
  
  
 
 Where to Go Next 
 
  
 
 •H.323-to-H.323 Connections on a Cisco Unified Border Element 
 
  
 
 •SIP-to-SIP Connections on a Cisco Unified Border Element 
 
  
 
 •Cisco Unified Border Element for H.323 Cisco Unified Communications Manager to H.323 Service Provider Connectivity 
 
  
 
 •Configuring Cisco Unified Border Element Videoconferencing 
 
   
  
  
 
 Additional References 
 
  
 
 The following sections provide references related to H.323-to-SIP IP-to-IP Gateway Connections 
 
  
 
 The following sections provide additional references related to the Cisco UBE Configuration Guide. 
 
 
 
   
 
 
 
 
  
 
 Note•In addition to the references listed below, each chapter provides additional references related to Cisco Unified Border Element. 
 
  
 
 •Some of the products and services mentioned in this guide may have reached end of life, end of sale, or both. Details are available at http://www.cisco.com/en/US/products/prod_end_of_life.html. 
 
  
 
 •The preface and glossary for the entire voice-configuration library suite of documents is listed below. 
 
 
 
  
   
  
 
 Related Documents
 
 
 
  
 
    
    
    
 
      
      

        Related Topic 
      
 		 
      
      

        Document Title 
      
 		 
    
 
    
 
     
 Cisco IOS commands 
 		 
     
 Cisco IOS Master Commands List, All Releases 
 		 
    
 
    
 
     
 Cisco IOS Voice commands 
 		 
     
 Cisco IOS Voice Command Reference 
 		 
    
 
    
 
     
 Cisco IOS Voice Configuration Library 
 		 
     
 For more information about Cisco IOS voice features, including feature documents, and troubleshooting information—at 
 
 http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/
cisco_ios_voice_configuration_library_glossary/vcl.htm 
 		 
    
 
    
 
     
 Cisco IOS Release 15.0 
 		 
     
 Cisco IOS Release 15.0 Configuration Guides  
 		 
    
 
    
 
     
 Cisco IOS Release 12.4 
 		 
     
 •Cisco IOS Release 12.4 Configuration Guides  
 
 •Cisco IOS Release 12.4T Configuration Guides  
 		 
    
 
    
 
     
 Cisco IOS Release 12.3 
 		 
     
 •Cisco IOS Release 12.3 documentation 
 
 •Cisco IOS Voice Troubleshooting and Monitoring Guide 
 
 •Tcl IVR Version 2.0 Programming Guide 
 		 
    
 
    
 
     
 Cisco IOS Release 12.2 
 		 
     
 Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 
 		 
    
 
    
 
     
 DSP documentation 
 		 
     
 High-Density Packet Voice Feature Card for Cisco AS5350XM and AS5400XM Universal Gateways 
 		 
    
 
    
 
     
 GKTMP (GK API) Documents 
 		 
     
 •GKTMP Command Reference 
 
 •GKTMP Messages:  
 		 
    
 
    
 
     
 internet Low Bitrate Codec (iLBC) Documents 
 		 
     
 •Codecs section of the Dial Peer Configuration on Voice Gateway Routers Guide 
 
 •Dial Peer Features and Configuration section of the Dial Peer Configuration on Voice Gateway Routers Guide 
 		 
    
 
    
 
     
 Cisco Unified Border Element Configuration Examples 
 		 
     
 •Local-to-remote network using the IPIPGW 
 
 •Remote-to-local network using the IPIPGW 
 
 •Remote-to-remote network using the IPIPGW 
 
 •Remote-to-remote network using two IPIPGWs 
 		 
    
 
    
 
     
 Related Application Guides 
 		 
     
 •Cisco Unified Communications Manager and Cisco IOS Interoperability Guide 
 
 •Cisco IOS Fax, Modem, and Text Support over IP Configuration Guide 
 
 •"Configuring T.38 Fax Relay" chapter 
 
 •Cisco IOS SIP Configuration Guide 
 
 •Cisco Unified Communications Manager (CallManager) Programming Guides 
 
 •Quality of Service for Voice over IP 
 		 
    
 
    
 
     
 Related Platform Documents 
 		 
     
 •Cisco 2600 Series Multiservice Platforms 
 
 •Cisco 2800 Series Integrated Services Routers  
 
 •Cisco 3600 Series Multiservice Platforms 
 
 •Cisco 3700 Series Multiservice Access Routers 
 
 •Cisco 3800 Series Integrated Services Routers 
 
 •Cisco 7200 Series Routers 
 
 •Cisco 7301 
 		 
    
 
    
 
     
 Related gateway configuration documentation 
 		 
     
 Media and Signaling Authentication and Encryption Feature for Cisco IOS H.323 Gateways. 
 		 
    
 
    
 
     
 Cisco IOS NAT Configuration Guide, Release 12.4T 
 		 
     
 Configuring Cisco IOS Hosted NAT Traversal for Session Border Controller 
 		 
    
 
    
 
     
 Troubleshooting and Debugging guides 
 		 
     
 •Cisco IOS Debug Command Reference, Release 12.4 
 
 •Troubleshooting and Debugging VoIP Call Basics 
 
 •VoIP Debug Commands 
 		 
    
 
    
  
 
 
 
 
 
 
 
 
   
  
 
 Standards
 
 
 
  
 
    
    
    
 
      
      

        Standard 
      
 		 
      
      

        Title 
      
 		 
    
 
    
 
     
 H.323 Version 4 and earlier 
 		 
     
 H.323 (ITU-T VOIP protocols) 
 		 
    
 
    
 
     
 H.323 - H.245 Version 12, Annex R 
 		 
     
 H.323 (ITU-T VOIP protocols) 
 		 
    
 
    
  
 
 
 
 
 
 
 
 
   
  
 
 MIBs
 
 
 
  
 
    
    
    
 
      
      

        MIB 
      
 		 
      
      

        MIBs Link 
      
 		 
    
 
    
 
     
 •CISCO-DSP-MGMT-MIB 
 
 •CISCO-VOICE-DIAL-CONTROL-MIB 
 
 •IP-TAP-MIB 
 
 •TAP2-MIB 
 
 •USER-CONNECTION-TAP-MIB 
 		 
     
 To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL: 
 
 http://www.cisco.com/go/mibs 
 		 
    
 
    
  
 
 
 
 
 
 
 
 
   
  
 
 RFCs
 
 
 
  
 
    
    
    
 
      
      

        RFC 
      
 		 
      
      

        Title 
      
 		 
    
 
    
 
     
 RFC 1889 
 		 
     
 RTP: A Transport Protocol for Real-Time Applications 
 		 
    
 
    
 
     
 RFC 2131 
 		 
     
 Dynamic Host Configuration Protocol 
 		 
    
 
    
 
     
 RFC 2132 
 		 
     
 DHCP Options and BOOTP Vendor Extensions 
 		 
    
 
    
 
     
 RFC 2833 
 		 
     
 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals 
 		 
    
 
    
 
     
 RFC 3203 
 		 
     
 DHCP reconfigure extension 
 		 
    
 
    
 
     
 RFC 3261 
 		 
     
 SIP: Session Initiation Protocol 
 		 
    
 
    
 
     
 RFC 3262 
 		 
     
 Reliability of Provisional Responses in Session Initiation Protocol (SIP) 
 		 
    
 
    
 
     
 RFC 3323 
 		 
     
 A Privacy Mechanism for the Session Initiation Protocol (SIP) 
 		 
    
 
    
 
     
 RFC 3325 
 		 
     
 Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks 
 		 
    
 
    
 
     
 RFC 3361 
 		 
     
 Dynamic Host Configuration Protocol (DHCP-for-IPv4) Option for Session Initiation Protocol (SIP) Servers 
 		 
    
 
    
 
     
 RFC 3455 
 		 
     
 Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP) 
 		 
    
 
    
 
     
 RFC 3608 
 		 
     
 Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration 
 		 
    
 
    
 
     
 RFC 3711 
 		 
     
 The Secure Real-time Transport Protocol (SRTP) 
 		 
    
 
    
 
     
 RFC 3925 
 		 
     
 Vendor-Identifying Vendor Options for Dynamic Host Configuration Protocol version 4 (DHCPv4) 
 		 
    
 
    
  
 
 
 
 
 
 
 
 
   
  
 
 Technical Assistance
 
 
 
  
 
    
    
    
 
      
      

        Description 
      
 		 
      
      

        Link 
      
 		 
    
 
    
 
     
 The Cisco Support and Documentation website provides online resources to download documentation, software, and tools. Use these resources to install and configure the software and to troubleshoot and resolve technical issues with Cisco products and technologies. Access to most tools on the Cisco Support and Documentation website requires a Cisco.com user ID and password. 
 		 
     
 http://www.cisco.com/cisco/web/support/index.html 
 		 
    
 
    
  
 
 
 
 
 
 
 
 
   
  
  
 
 Feature Information for H.323-to-SIP Connections on a Cisco Unified Border Element 
 
  
 
 Table 1 lists the features in this module and provides links to specific configuration information. Only features that were introduced or modified in Cisco IOS Release 12.3(1) or a later release appear in the table. 
 
  
 
 For information on a feature in this technology that is not documented here, see the "Cisco Unified Border Element Features Roadmap." 
 
  
 
 Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required. 
 
 
 
   
 
 
 
 
  
 
 Note Table 1 lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature. 
 
 
 
  
 
 
 
 
 
  
 
    
     
     
    
 Table 1 Feature Information for H.323-to-SIP Connections on a Cisco Unified Border Element  
 
    
    
    
 
      
      

        Feature Name 
      
 		 
      
      

        Releases 
      
 		 
      
      

        Feature Information 
      
 		 
    
 
    
 
     
 Accounting 
 		 
     
 12.3(11)T 
 		 
     
 RADIUS call-accounting records, calling/called name and number. 
 		 
    
 
    
 
     
 Call Admission Control 
 		 
     
 12.3(11)T 
 		 
     
 RSVP synchronized with call signaling. 
 		 
    
 
    
 
     
 Cisco Unified Communications Manager Connections 
 		 
     
 12.4(6)XE 
 		 
     
 H.323-to-SIP Supplementary services for Cisco Unified Communications Manager with MTP on the H.323 Trunk 
 		 
    
 
    
 
     
 Cisco UBE MIB support 
 		 
     
 15.0(1)XA 
 		 
     
 This feature was introduced. 
 		 
    
 
    
 
     
 Codec Support 
 		 
     
 12.4(11)T 
 		 
     
 iLBC Codec Support 
 		 
    
 
    
 
     
 Codec Transcoding 
 		 
     
 12.3(11)T 
 		 
     
 Codec transcoding (G.711-G.729)—This feature enables the IP-to-IP gateway to bridge calls between networks that support different VoIP call-signaling protocols (SIP and H.323) 
 		 
    
 
    
 
     
 DTMF 
 		 
     
 12.3(11)T 
 
 12.4(6)XE 
 		 
     
 12.3(11)T—DTMF relay 
 
 •H.245 alpha/signal <--> SIP RFC 2833 
 
 •H.245 alpha/signal <--> SIP Notify 
 
 12.4(6)XE—G.711 Inband DTMF to RFC 2833 
 		 
    
 
    
 
     
 Fax/Modem 
 		 
     
 12.3(11)T 
 		 
     
 T.38 fax relay and Cisco fax relay 
 		 
    
 
    
 
     
 Interworking Between RSVP Capable and RSVP Incapable Networks 
 		 
     
 15.0(1)XA
15.1(3)T 
 		 
     
 The Support for Interworking Between RSVP Capable and RSVP Incapable Networks feature provides precondition-based RSVP support for basic audio call and supplementary services on the Cisco UBE. 
 
 The following section provides information about this feature: 
 
 •Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks 
 
 15.1(3)T—Configuring EO-EO, DO-DO and DO-EO support on dial peer. 
 		 
    
 
    
 
     
 Managing H.323 IP Group Call Capacities 
 		 
     
 12.2(13)T 
 		 
     
 Creates a maximum capacity for the IP group providing extra control for load and resource balancing. 
 		 
    
 
    
 
     
 Mapping ECS to ReINVITE and ECS to REFER on the Cisco IOS SBC. 
 		 
     
 12.4(20)T 
 		 
     
 H.323-to-SIP Supplementary Service Enhancements for Session Border Controller (SBC) 
 		 
    
 
    
 
     
 Media Modes 
 		 
     
 12.3(1) 
 		 
     
 Media flow-around capability on the IP-to-IP gateway by supporting the processing of call set-up and teardown request (VoIP call signaling) and for media streams (flow-through and flow-around) 
 		 
    
 
    
 
     
 Rotary Support 
 		 
     
 12.3(11)T 
 		 
     
 H.323-to-H.323 Call Failure Recovery (Rotary) on a Cisco Unified Border Element. Eliminates codec restrictions and enables the Cisco UBE to restart codec negotiation with the originating endpoint based on the codec capabilities of the next dial peer in the rotary group for H.323-to-H.323 interconnections. 
 		 
    
 
    
 
     
 Signaling Interworking 
 		 
     
 12.3(11)T
12.4(4)T 
 		 
     
 12.3(11)T—This feature enables SIP-to-H.323 protocol interworking capabilities of the Cisco Unified Border Element: 
 
 •Interworking between H.323 Fast-Start and SIP early-media signaling 
 
 •Interworking between H.323 Slow-Start and SIP delayed-media signaling 
 
 12.4(4)T—Extended SIP-to-H.323 Call Interworking for Session Border Controller (SBC) 
 		 
    
 
    
 
     
 TCL IVR 
 		 
     
 12.3(11)T
12.4(11)T 
 		 
     
 12.3(11)T—TCL IVR 2.0 for SIP, including media playout and digit collection (RFC 2833 DTMF relay) 
 
 12.4(11)T —TCL IVR support with SIP NOTIFY DTMF 
 		 
    
 
    
 
     
 Transport Protocols 
 		 
     
 12.3(11)T 
 		 
     
 UDP and TCP transport 
 		 
    
 
    
 
     
 VXML 
 		 
     
 12.4(6)XE
12.4(11)T 
 		 
     
 12.4(6)XE— 
 
 •VXML 3.x support 
 
 •VXML support with SIP Notify 
 
 12.4(11)T—VXML support with SIP NOTIFY DTMF 
 		 
    
 
    
  
 
 
 
 
 
 
 
 
 
 
  
  
  
 
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