- Cisco Unified Border Element Protocol-Independent Features and Setup
- Interworking Between RSVP Capable and RSVP Incapable Networks
- SIP INFO Method for DTMF Tone Generation
- WebEx Telepresence Media Support Over Single SIP Session
- DTMF Events through SIP Signaling
- Call Progress Analysis Over IP-to-IP Media Session
- Codec Preference Lists
- AAC-LD MP4A-LATM Codec Support on Cisco UBE
- Multicast Music-on-Hold Support on Cisco UBE
- Network-Based Recording
- Video Recording - Additional Configurations
- TDoS Attack Mitigation
- Cisco Unified Communications Gateway Services--Extended Media Forking
- Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls
- Acoustic Shock Protection
- Noise Reduction
- iLBC Support for SIP and H.323
- Configuring RTP Media Loopback for SIP Calls
- SIP Ability to Send a SIP Registration Message on a Border Element
- Session Refresh with Reinvites
- SIP Stack Portability
- Interworking of Secure RTP calls for SIP and H.323
- Cisco UBE Support for SRTP-RTP Internetworking
- Support for SRTP Termination
- Configuring RTCP Report Generation
- SIP SRTP Fallback to Nonsecure RTP
- Configuring Support for Interworking Between RSVP Capable and RSVP Incapable Networks
- VoIP for IPv6
- Mid-call Signaling Consumption
- Support for Software Media Termination Point
- Cisco Unified Communication Trusted Firewall Control
- Cisco Unified Communication Trusted Firewall Control-Version II
- Domain-Based Routing Support on the Cisco UBE
- URI-Based Dialing Enhancements
- Fax Detection for SIP Call and Transfer
Configuring RTP Media Loopback for SIP Calls
RTP packets are looped back toward the source device when the RTP Media Loopback for SIP Calls feature is configured on a dial peer. The SIP RTP media loopback can be used during Cisco UBE deployments to make test calls to verify the media path between the endpoints and Cisco UBE. In a voice loopback call, an echo is heard at the device originating the call. In a video loopback call, the locally captured video and the audio echo must be rendered at the source device.
- Finding Feature Information
- Prerequisites
- Restrictions
- Information About RTP Media Loopback for SIP Calls
- How to Configure RTP Media Loopback for SIP Calls
- Configuration Examples for RTP Media Loopback
- Feature Information for RTP Media Loopback for SIP Calls
Finding Feature Information
Your software release may not support all the features documented in this module. For the latest caveats and feature information, see Bug Search Tool and 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 table.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Prerequisites
Use the media flow-through command in dial peer voice or voice service configuration mode to enable the media packets.
Cisco Unified Border Element
Cisco IOS Release 15.1(4)M or a later release must be installed and running on your Cisco Unified Border Element.
Cisco Unified Border Element (Enterprise)
Restrictions
Information About RTP Media Loopback for SIP Calls
Digital Signal Processors (DSP) generate and transmit Real-time Transport Protocol (RTP) media packets from a source to a destination transport address during a SIP call session. However, when a SIP call is put on hold the DSP stops generating the RTP media packets and resumes generating and transmitting these media packets after the SIP call is resumed. This ensures that the RTP sequence number is continuous from the time of the origin to the end of a SIP call.
How to Configure RTP Media Loopback for SIP Calls
RTP packets are looped back toward the source device when the RTP Media Loopback for SIP Calls feature is configured on a dial peer. Perform this task to enable the RTP Media Loopback for SIP Calls feature on a dial peer.
1.
enable
2.
configure
terminal
3.
dial-peer
voice
tag
voip
4.
destination-pattern
string
5.
session
protocol
sipv2
6.
session
target
loopback:rtp
7.
incoming
called-number
string
8.
exit
DETAILED STEPS
Configuration Examples for RTP Media Loopback
- Example: Configuring Video Loopback with Cisco Telepresence System
- Example: Configuring Video Loopback with Cisco Unified Video Advantage
Example: Configuring Video Loopback with Cisco Telepresence System
The following sample output shows Media Loopback for SIP Calls configured on a Cisco Telepresence System (CTS).
! codec profile 1 aacld fmtp "fmtp:96 profile-level-id=16;streamtype=5;mode=AAChbr;config=B98C00;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDura tion=480" ! codec profile 2 h264 fmtp "fmtp:112 profile-level-id=4D0028;sprop-parametersets= R00AKAmWUgDwBDyA,SGE7jyA=;packetization-mode=1" ! voice class codec 4 codec preference 1 aacld profile 1 video codec h264 profile 2 ! dial-peer voice 2000 voip destination-pattern 2000 rtp payload-type cisco-codec-fax-ind 110 rtp payload-type cisco-codec-aacld 96 rtp payload-type cisco-codec-video-h264 112 session protocol sipv2 session target loopback:rtp incoming called-number 2000 voice-class codec 4 voice-class sip bandwidth audio tias-modifier 64000 voice-class sip bandwidth video tias-modifier 4500000 !
Example: Configuring Video Loopback with Cisco Unified Video Advantage
The following sample output shows Media Loopback for SIP Calls configured on a Cisco Unified Video Advantage (CUVA).
! codec profile 3 h264 fmtp "fmtp:98 profile-level-id=420015" ! voice class codec 6 codec preference 1 g711ulaw video codec h264 profile 3 ! dial-peer voice 5000 voip description CUVA destination-pattern 5000 rtp payload-type cisco-codec-video-h264 98 session protocol sipv2 session target loopback:rtp incoming called-number 5000 voice-class codec 6 voice-class sip bandwidth video tias-modifier 384000
Feature Information for RTP Media Loopback for SIP Calls
The following table provides release information about the feature or features described in this module. This table 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.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to . An account on Cisco.com is not required.Feature History Table entry for the Cisco Unified Border Element.
Feature Name |
Releases |
Feature Information |
---|---|---|
RTP Media Loopback for SIP Calls |
15.1(4)M 15.2(1)T |
RTP packets are looped back toward the source when the RTP Media Loopback for SIP Calls feature is configured on a dial peer. SIP RTP media loopback helps in verifying the media path between the device originating the call and the intermediate device. The following commands were introduced or modified: None. |