SIP-SIP VideoDelayed Offer-Delayed Offer

(Pina Martini Phase 1)

Topology

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 Video Call with CTS/CUCMCUBECTS/CUCM CUCM/CTS

INVITE (w/o SDP)

100 TryingINVITE (w/o SDP)

100 Trying

180 Ringing180 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/CUCMCUBECUVA/CUCM CUCM/CUVA

INVITE (w/o SDP)

100 TryingINVITE (w/o SDP)

100 Trying

180 Ringing180 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)

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 (SDP: m-line audio, m-line video, a=recvonly)

INVITE (SDP: m-line audio, m-line video, a=sendrecv)

ACK

ACK

200 OK (SDP: m-line audio, m-line video, a=sendrecv)

Audio/Video Media Path

SIP - SIP VideoDelayed 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/CUCMCUBECTS/CUCM CUCM/CTS

INVITE (w/o SDP)

100 TryingINVITE (SDP: m-line audio, m-line video, a=sendrecv)

100 Trying

180 Ringing180 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

Audio/Video Media Path

SDP generated from codec list configured under dial-

peer.

ACK does not contain SDP, as Offer – Answer is already

completed.

SIP-SIP DO-EO Video Call with CUVA/CUCMCUBECUVA/CUCM CUCM/CUVA

INVITE (w/o SDP)

100 TryingINVITE (SDP: m-line audo, m-line video, a=sendrecv)

100 Trying

180 Ringing180 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)

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 (w/o SDP)

INVITE (SDP: m-line audio, m-line video, a=sendrecv)

ACK

ACK

200 OK (SDP: m-line audio, m-line video, a=sendrecv)

Audio/Video Media Path

SIP - SIP VideoEarly Offer – Early Offer

(Pina Martini Phase 2)

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/CUCMO-CUBECTS/CUCM CUCM/CTS

INVITE (w/o SDP)

100 Trying

INVITE (SDP: m-line audio, video)100 Trying

180 Ringing

180 Ringing183 Progress (SDP: m-line audio, video)

200 OK (SDP: m-line audio, video)

ACK (SDP: m-line audio, video)ACK (w/o SDP)

Audio/Video Media Path

DO – EO configured at Originating CUBE .

T-CUBE

EO – EO call for Terminating CUBE .

INVITE (SDP: m-line audio, video)

100 Trying

180 Ringing

183 Progress (SDP: m-line audio, video)

183 (SDP: m-line audio, video)

200 OK (SDP: m-line audio, video)

200 (SDP: m-line audio, video)

ACK

SIP - SIP VideoFlow 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 AroundCUBECTS/CUCM CUCM/CTS

INVITE (w/o SDP)

100 TryingINVITE (w/o SDP)

100 Trying

180 Ringing180 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)

SDP carries port and address of terminating CUCM/CTS

SDP carries port and address of terminating CUCM/CTS

SDP carries port and address of originating CUCM/CTS

Audio/Video Media Path directly between CUCM/CTS

Configuration• The CLI for DO-EO can configured either under global or at the outgoing

dial-peerIn 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 followsdial-peer voice 1 voipvoice-class sip early-offer forced

• The system keyword can be used in the dial-peer level to enable the global settingdial-peer voice 1 voipvoice-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 • 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 voipcodec aacld profile 1video codec h264 profile 2

Sample Configuration

voice service voip media flow-around

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 sipv2codec transparent

dial-peer voice 11 voip destination-pattern 7005 media flow-around rtp payload-type cisco-codec-fax-ind 120 rtp payload-type cisco-codec-aacld 96 session protocol sipv2 session target ipv4:9.13.2.90 codec transparent

Global Configuration

OR

Dialpeer Configurationvoice 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 sipv2codec transparent

dial-peer voice 11 voip destination-pattern 7005 voice-class media 1 rtp payload-type cisco-codec-fax-ind 120 rtp payload-type cisco-codec-aacld 96 session protocol sipv2 session target ipv4:9.13.2.90 codec transparent

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 VideoReceived: 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-269572To: <sip:7005@9.13.29.23> CSeq: 101 INVITEExpires: 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=z9hFrom: <sip:7000@9.13.29.23>;tag=38398BF4-F07 To: <sip:7005@9.13.2.90>;tag=0a98a44b-1efa-4b64-90b6-58e24675e1d8-30105493

v=0o=CiscoSystemsCCM-SIP 2000 1 IN IP4 9.13.2.90 s=SIP Callc=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=480a=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=z9hG4bK13From: <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 9.13.29.23s=SIP Call c=IN IP4 9.13.29.23t=0 0m=audio 19092 RTP/AVP 0c=IN IP4 9.13.29.23a=rtpmap:0 PCMU/8000a=ptime:20 m=video 17512 RTP/AVP 112 b=TIAS:4000000a=rtpmap:112 H264/90000 a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1

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-4To: <sip:7005@9.13.29.23>;tag=38398C70-310 Date: Tue, 17 Jul 2007 05:12:57 GMT CSeq: 101 ACK v=0o=CiscoSystemsCCM-SIP 2000 1 IN IP4 9.13.25.50 s=SIP Callc=IN IP4 9.13.29.37 t=0 0 m=audio 16384 RTP/AVP 0b=TIAS:64000a=rtpmap:0 PCMU/8000a=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/2.0Via: SIP/2.0/UDP 9.13.29.23:5060;branch=z9hG4bK48CB1From: <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 c=IN IP4 9.13.29.23 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 16730 RTP/AVP 112 c=IN IP4 9.13.29.23b=TIAS:4000000 a=rtpmap:112 H264/90000a=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/2.0Via: SIP/2.0/UDP 9.13.29.23:5060;branch=z9hG4bK18D28From: <sip:7000@9.13.29.23>;tag=439D94-DCFTo: <sip:7005@9.13.2.90>Call-ID: 90E3747-ABC111DC-8040E-416623EF@9.13.29.23CSeq: 101 INVITEContact: <sip:7000@9.13.29.23:5060>Content-Length: 551

v=0o=CiscoSystemsSIP-GW-UserAgent 5305 1760 IN IP4 9.13.24.100s=SIP Callc=IN IP4 9.13.24.100t=0 0m=audio 17270 RTP/AVP 96 0 19c=IN IP4 9.13.24.100a=rtpmap:96 mpeg4-generic/48000a=fmtp:96 profile-level-id=16;streamtype=5;config=11B0;mode=AAC-hbr;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDuration=480a=rtpmap:0 PCMU/8000a=rtpmap:19 CN/8000m=video 18800 RTP/AVP 112c=IN IP4 9.13.24.100a=rtpmap:112 H264/90000a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1

Debugs• SDP info. in incoming 183/200OK

v=0o=CiscoSystemsCCM-GW-UserAgent 2071 6347 IN IP4 9.13.2.90

s=SIP Callc=IN IP4 9.13.29.80t=0 0m=audio 16384 RTP/AVP 96 0 18 19 c=IN IP4 9.13.29.80a=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 9.13.8.90s=SIP Callc=IN IP4 9.13.29.80t=0 0m=audio 16384 RTP/AVP 96 0 18 19 c=IN IP4 9.13.29.80a=rtpmap::96 mpeg4-generic/48000 a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:19 CN/8000m=video 16388 RTP/AVP 112c=IN IP4 9.13.29.80b=TIAS:4000000a=rtpmap:112 H264/90000

video-ipipgw1#sh voip rtp connections No active connections found

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

Additional Information