SIP-SIP Video Delayed Offer-Delayed Offer (Pina Martini Phase 1)
Topology Note: End point shown in diagram is SIP Phone or SCCP Phone
Support for SIP-SIP Delay Offer to Delay Offer SIP-SIP Video Support for SIP-SIP Delay Offer to Delay Offer SIP Endpoints supported: Cisco TelePresence (CTS), CUVA Video Codecs supported: H264, H263 “codec transparent” configuration required for incoming/outgoing dialpeers for video calls SIP-SIP “codec transparent” support is for video only.
SIP-SIP Video Call with CTS/CUCM CUBE CUCM/CTS INVITE (w/o SDP) 100 Trying INVITE (w/o SDP) 100 Trying 180 Ringing 180 Ringing 183 Progress (SDP: m-line audio, m-line video, a=sendrecv) 183 Progress (SDP: m-line audio, m-line video, a=sendrecv) 200 OK (SDP: m-line audio, m-line video, a=sendrecv) 200 OK (SDP: m-line audio, m-line video, a=sendrecv) ACK (SDP: m-line audio, m-line video, a=sendrecv) ACK (SDP: m-line audio, m-line video, a=sendrecv) Audio/Video Media Path
SIP-SIP Video Call with CUVA/CUCM CUBE CUCM/CUVA INVITE (w/o SDP) 100 Trying INVITE (w/o SDP) 100 Trying 180 Ringing 180 Ringing 183 Progress (SDP: m-line audio, m-line video, a=sendonly) 183 Progress (SDP: m-line audio, m-line video, a=sendonly) 200 OK (SDP: m-line audio, m-line video, a=sendonly) 200 OK (SDP: m-line audio, m-line video, a=sendonly) ACK (SDP: m-line audio, m-line video, a=recvonly) ACK (SDP: m-line audio, m-line video, a=recvonly) INVITE (SDP: m-line audio, m-line video, a=sendrecv) INVITE (SDP: m-line audio, m-line video, a=sendrecv) 100 Trying 100 Trying 200 OK (SDP: m-line audio, m-line video, a=sendrecv) ACK 200 OK (SDP: m-line audio, m-line video, a=sendrecv) ACK Audio/Video Media Path
SIP - SIP Video Delayed Offer – Early Offer (Pina Martini Phase 2)
DO-EO SIP-SIP Video CUBE generates an outgoing Early Offer INVITE with the configured codec list, for a incoming Delayed Offer INVITE The CLI developed for DO-EO Audio is re-used to enable DO-EO Video. DO-EO Audio call if only audio codecs are configured under dial-peer. DO-EO Video call if both audio and video codecs are configured under dial-peer. A New CLI “codec-profile” added to define codec attributes for Video (H263, H264) and Audio (AACLD) codecs. The codec attributes configured under codec-profile is used to generate the a=fmtp attribute line in the Early Offer SDP
SIP-SIP DO-EO Video Call with CTS/CUCM CUBE SDP generated from codec list configured under dial-peer. CUCM/CTS INVITE (w/o SDP) 100 Trying INVITE (SDP: m-line audio, m-line video, a=sendrecv) 100 Trying 180 Ringing 180 Ringing 183 Progress (SDP: m-line audio, m-line video, a=sendrecv) 183 Progress (SDP: m-line audio, m-line video, a=sendrecv) 200 OK (SDP: m-line audio, m-line video, a=sendrecv) 200 OK (SDP: m-line audio, m-line video, a=sendrecv) ACK does not contain SDP, as Offer – Answer is already completed. ACK (SDP: m-line audio, m-line video, a=sendrecv) ACK Audio/Video Media Path
SIP-SIP DO-EO Video Call with CUVA/CUCM CUBE CUCM/CUVA INVITE (w/o SDP) 100 Trying INVITE (SDP: m-line audo, m-line video, a=sendrecv) 100 Trying 180 Ringing 180 Ringing 183 Progress (SDP: m-line audio, m-line video, a=sendonly) 183 Progress (SDP: m-line audio, m-line video, a=sendonly) 200 OK (SDP: m-line audio, m-line video, a=sendonly) 200 OK (SDP: m-line audio, m-line video, a=sendonly) ACK (SDP: m-line audio, m-line video, a=recvonly) ACK (w/o SDP) INVITE (SDP: m-line audio, m-line video, a=sendrecv) 100 Trying INVITE (SDP: m-line audio, m-line video, a=sendrecv) 100 Trying 200 OK (SDP: m-line audio, m-line video, a=sendrecv) ACK 200 OK (SDP: m-line audio, m-line video, a=sendrecv) ACK Audio/Video Media Path
SIP - SIP Video Early Offer – Early Offer (Pina Martini Phase 2)
No CLI configuration required to support EO-EO. SIP-SIP EO-EO Video CUBE supports receiving an Early Offer INVITE with both audio and video m lines. CUBE will always generate an outgoing Early Offer INVITE when in response to an incoming Early Offer INVITE. EO-DO is not supported. No CLI configuration required to support EO-EO.
SIP-SIP EO-EO Video Call with CTS/CUCM O-CUBE T-CUBE CUCM/CTS DO – EO configured at Originating CUBE . INVITE (w/o SDP) EO – EO call for Terminating CUBE . 100 Trying INVITE (SDP: m-line audio, video) 100 Trying INVITE (SDP: m-line audio, video) 100 Trying 180 Ringing 180 Ringing 180 Ringing 183 Progress (SDP: m-line audio, video) 183 Progress (SDP: m-line audio, video) 183 (SDP: m-line audio, video) 200 OK (SDP: m-line audio, video) 200 OK (SDP: m-line audio, video) 200 (SDP: m-line audio, video) ACK (SDP: m-line audio, video) ACK (w/o SDP) ACK Audio/Video Media Path
SIP - SIP Video Flow Around Pina Martini (Phase 2)
SIP-SIP Video Flowaround Flow Around Supported for DO-DO, EO-EO Calls. DO-EO not supported. Feature enabled by existing CLI used for Audio Flowaround. Only SIP Signaling done via CUBE, while media flows end to end. No media termination in CUBE for both Audio and Video Streams. CUBE will pass through the endpoint media address and port in received in SDP between the in and out legs.
SIP-SIP Video Flowaround Flow Around Supported for DO-DO, EO-EO Calls. DO-EO not supported. Feature enabled by existing CLI used for Audio Flowaround. Only SIP Signaling done via CUBE, while media flows end to end. No media termination in CUBE for both Audio and Video Streams. CUBE will pass through the endpoint media address and port in received in SDP between the in and out legs.
SIP-SIP Video Flow Around CTS/CUCM CUBE CUCM/CTS INVITE (w/o SDP) 100 Trying INVITE (w/o SDP) 100 Trying 180 Ringing 180 Ringing SDP carries port and address of terminating CUCM/CTS 183 Progress (SDP: m-line audio, m-line video, a=sendrecv) 183 Progress (SDP: m-line audio, m-line video, a=sendrecv) 200 OK (SDP: m-line audio, m-line video, a=sendrecv) SDP carries port and address of terminating CUCM/CTS 200 OK (SDP: m-line audio, m-line video, a=sendrecv) SDP carries port and address of originating CUCM/CTS ACK (SDP: m-line audio, m-line video, a=sendrecv) ACK (SDP: m-line audio, m-line video, a=sendrecv) Audio/Video Media Path directly between CUCM/CTS
Configuration voice service voip sip early-offer forced The CLI for DO-EO can configured either under global or at the outgoing dial-peer In the global level, the CLI has to be configured under voice service voip sip as follows voice service voip sip early-offer forced In the dial-peer level, the CLI has to be configured on the outgoing dial-peer as follows dial-peer voice 1 voip voice-class sip early-offer forced The system keyword can be used in the dial-peer level to enable the global setting voice-class sip early-offer forced system The early-offer forced CLI is disabled by default
Configuration - cont Codecs should be configured on the CUBE outleg so that it can send them out in Early-Offer, codec T on the outleg will not send all the media parameters that are configured Since CTS uses 96 as default payload-type for AAC-LD and 112 for H.264 codecs, it is necessary to change the default IOS fax payload-type from 96 to another value and assign 96 as the payload-type for AAC-LD and 112 for H.264 codecs. dial-peer voice 1 voip rtp payload-type cisco-codec-fax-ind 120 rtp payload-type cisco-codec-aacld 96 rtp payload-type cisco-codec-video-h264 112 session protocol sipv2 incoming called-number 7005
Configuration - cont fmtp config: codec profile 1 aacld The fmtp parameters can be configured on the codec profile and applied to the outgoing dial-peer as follows: fmtp config: codec profile 1 aacld fmtp "fmtp:96 profile-level-id=16;streamtype=5;config=11B0;mode=AAC-hbr;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDuration=500“ codec profile 2 h264 fmtp "fmtp:112 profile-level-id=DEFGHI;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1“ Dial-peer config: dial-peer voice 1 voip codec aacld profile 1 video codec h264 profile 2
Sample Configuration Global Configuration Dialpeer Configuration voice service voip media flow-around Dialpeer Configuration voice class media 1 media flow-around dial-peer voice 10 voip voice-class media 1 rtp payload-type cisco-codec-fax-ind 120 rtp payload-type cisco-codec-aacld 96 session protocol sipv2 codec transparent dial-peer voice 11 voip destination-pattern 7005 session target ipv4:9.13.2.90 dial-peer voice 10 voip media flow-around rtp payload-type cisco-codec-fax-ind 120 rtp payload-type cisco-codec-aacld 96 session protocol sipv2 codec transparent dial-peer voice 11 voip destination-pattern 7005 session target ipv4:9.13.2.90 OR
Debugs & Show commands Show voip rtp connections shows no RTP stream for FA session Show call active video shows if the call is flow-around or flow-thru in “Media Setting” field Debug ccsip all
Debugs for SIP-SIP DO-DO Video Received: INVITE sip:7005@9.13.29.23:5060 SIP/2.0 Via: SIP/2.0/TCP 9.13.25.50;branch=z9hG1d57 Remote-Party-ID: <sip:7000@9.13.25.50>;party=calling;screen=yes;privacy=off From: <sip:7000@9.13.25.50>;tag=873d105b-74ee-43aa-a354-fb6b50eddeba-269572 To: <sip:7005@9.13.29.23> CSeq: 101 INVITE Expires: 60 Contact: <sip:7000@9.13.25.50:5060;transport=tcp>;video;audio Max-Forwards: 69 Content-Length: 0 Sent: INVITE sip:7005@9.13.2.90:5060 SIP/2.0 Via: SIP/2.0/UDP 9.13.29.23:5060;branch=z9hG4bK Remote-Party-ID: <sip:7000@9.13.29.23>;party=calling;screen=yes;privacy=off From: <sip:7000@9.13.29.23>;tag=38398BF4-F07 To: <sip:7005@9.13.2.90> CSeq: 101 INVITE Expires: 180 Contact: <sip:7000@9.13.29.23:5060> Max-Forwards: 68 Content-Length: 0
Debugs for SIP-SIP Video Contd … Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 9.13.29.23:5060;branch=z9h From: <sip:7000@9.13.29.23>;tag=38398BF4-F07 To: <sip:7005@9.13.2.90>;tag=0a98a44b-1efa-4b64-90b6-58e24675e1d8-30105493 v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 9.13.2.90 s=SIP Call c=IN IP4 9.13.29.80 t=0 0 m=audio 16384 RTP/AVP 96 0 101 b=TIAS:64000 a=rtpmap:96 mpeg4-generic/48000 a=fmtp:96 profile-level-id=16;streamtype=5;config=11B0;mode=AAC-hbr;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDuration=480 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 16388 RTP/AVP 112 b=TIAS:4000000 a=rtpmap:112 H264/90000 a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1 Sent: SIP/2.0 200 OK Via: SIP/2.0/TCP 9.13.25.50;branch=z9hG4bK13 From: <sip:7000@9.13.25.50>;tag=873d105b-74ee-43aa-a354-fb6b50eddeba-26957231 To: <sip:7005@9.13.29.23>;tag=38398C70-310 CSeq: 101INVITE v=0 o=CiscoSystemsSIP-GW-UserAgent 8120 1880 IN IP4 9.13.29.23 s=SIP Call c=IN IP4 9.13.29.23 t=0 0 m=audio 19092 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 17512 RTP/AVP 112 b=TIAS:4000000 a=rtpmap:112 H264/90000 a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1 AAC_LD will be supported in PI1 Gold. Traces are not available as of now. So the above debugs will change (IPIPGW will pass the fmtp parameters for AAC_LD.
Debugs for SIP-SIP Video Contd … Received: ACK sip:7005@9.13.29.23:5060;transport=tcp SIP/2.0 \ Via: SIP/2.0/TCP 9.13.25.50;branch=z9hG4bK1384b1 From: <sip:7000@9.13.25.50>;tag=873d105b-74ee-4 To: <sip:7005@9.13.29.23>;tag=38398C70-310 Date: Tue, 17 Jul 2007 05:12:57 GMT CSeq: 101 ACK v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 9.13.25.50 s=SIP Call c=IN IP4 9.13.29.37 t=0 0 m=audio 16384 RTP/AVP 0 b=TIAS:64000 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 16388 RTP/AVP 112 b=TIAS:4000000 a=rtpmap:112 H264/90000 a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1 Sent: ACK sip:7005@9.13.2.90:5060 SIP/2.0 Via: SIP/2.0/UDP 9.13.29.23:5060;branch=z9hG4bK48CB1 From: <sip:7000@9.13.29.23>;tag=38398BF4-F07 To: <sip:7005@9.13.2.90>;tag=0a98a44b-1efa-4b64-90b6-58e24675e1d8-30105493 CSeq: 101 ACK v=0 o=CiscoSystemsSIP-GW-UserAgent 6986 3015 IN IP4 9.13.29.23 s=SIP Call c=IN IP4 9.13.29.23 t=0 0 m=audio 19528 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 16730 RTP/AVP 112 b=TIAS:4000000 a=rtpmap:112 H264/90000 a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1
Debugs for a basic DO-EO video call Received: INVITE sip:7005@9.13.29.23:5060 SIP/2.0 Via: SIP/2.0/TCP 9.13.25.50;branch=z9hG4bK From: <sip:7000@9.13.25.50>;tag=ef8f6d7d-4990-46d3-9ee9-fc3f982088f5-17247359 To: <sip:7005@9.13.29.23> Call-ID: f30b500-76614fbb-48a385-32190d09@9.13.25.50 Supported: timer,replaces Min-SE: 1800 CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Call-Info: <sip:9.13.25.50:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Session-Expires: 1800 Contact: <sip:7000@9.13.25.50:5060;transport=tcp>;video;audio Max-Forwards: 69 Content-Length: 0 Sent: INVITE sip:7005@9.13.2.90:5060 SIP/2.0 Via: SIP/2.0/UDP 9.13.29.23:5060;branch=z9hG4bK18D28 From: <sip:7000@9.13.29.23>;tag=439D94-DCF To: <sip:7005@9.13.2.90> Call-ID: 90E3747-ABC111DC-8040E-416623EF@9.13.29.23 CSeq: 101 INVITE Contact: <sip:7000@9.13.29.23:5060> Content-Length: 551 v=0 o=CiscoSystemsSIP-GW-UserAgent 5305 1760 IN IP4 9.13.24.100 s=SIP Call c=IN IP4 9.13.24.100 t=0 0 m=audio 17270 RTP/AVP 96 0 19 a=rtpmap:96 mpeg4-generic/48000 a=fmtp:96 profile-level-id=16;streamtype=5;config=11B0;mode=AAC-hbr;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDuration=480 a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 m=video 18800 RTP/AVP 112 a=rtpmap:112 H264/90000 a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1
Debugs video-ipipgw1#sh voip rtp connections SDP info. in incoming 183/200OK v=0 o=CiscoSystemsCCM-GW-UserAgent 2071 6347 IN IP4 9.13.2.90 s=SIP Call c=IN IP4 9.13.29.80 t=0 0 m=audio 16384 RTP/AVP 96 0 18 19 c=IN IP4 9.13.29.80 a=rtpmap::96 mpeg4-generic/48000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:19 CN/8000 m=video 16388 RTP/AVP 112 c=IN IP4 9.13.29.80 b=TIAS:4000000 a=rtpmap:112 H264/90000 SDP info. in outgoing 183/200OK v=0 o=CiscoSystemsCCM-GW-UserAgent 2003 2524 IN IP4 9.13.8.90 s=SIP Call c=IN IP4 9.13.29.80 t=0 0 m=audio 16384 RTP/AVP 96 0 18 19 c=IN IP4 9.13.29.80 a=rtpmap::96 mpeg4-generic/48000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:19 CN/8000 m=video 16388 RTP/AVP 112 c=IN IP4 9.13.29.80 b=TIAS:4000000 a=rtpmap:112 H264/90000 video-ipipgw1#sh voip rtp connections No active connections found
Additional Information AAC_LD Codec from CTS using Payload type 96 (which is reserved for FAX). So need to change the payload type foe FAX to some other value. 'Sh call active video compact' shows audio codec for a Video call (CSCsj68008). However ‘Sh call active video’ shows correct value in ‘VideoCap_Codec’ field Video codecs supported are H263 and H264. All audio codecs supported including AAC_LD Only Hold/Resume of Video calls supported. When CTS does RTP/RTCP multiplexing, Media Inactivity based on RTCP will not work