1. SIP-SIP Video Delayed Offer-Delayed Offer
(Pina Martini Phase 1)

2. Topology
Note: End point shown in diagram is SIP Phone or SCCP Phone

3. 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.

4. 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

5. 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

6. SIP - SIP Video
Delayed Offer – Early Offer
(Pina Martini Phase 2)

7. 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

8. 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

9. 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

10. SIP - SIP Video
Early Offer – Early Offer
(Pina Martini Phase 2)

11. 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.

12. 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

13. SIP - SIP Video
Flow Around
Pina Martini (Phase 2)

14. 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.

15. 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.

16. 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

17. 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

18. 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-h
session protocol sipv2
incoming called-number 7005

19. 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

20. 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:
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: OR

21. 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

22. Debugs for SIP-SIP DO-DO Video
Received:
INVITE SIP/2.0
Via: SIP/2.0/TCP ;branch=z9hG1d57
Remote-Party-ID:
From:
To:
CSeq: 101 INVITE
Expires: 60
Contact:
Max-Forwards: 69
Content-Length: 0
Sent:
INVITE SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK
Remote-Party-ID:
From:
To:
CSeq: 101 INVITE
Expires: 180
Contact:
Max-Forwards: 68
Content-Length: 0

23. Debugs for SIP-SIP Video Contd …
Received:
SIP/ OK
Via: SIP/2.0/UDP :5060;branch=z9h
From:
To:
v=0
o=CiscoSystemsCCM-SIP IN IP
s=SIP Call
c=IN IP
t=0 0
m=audio RTP/AVP
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:
m=video RTP/AVP 112
b=TIAS:
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1
Sent:
SIP/ OK
Via: SIP/2.0/TCP ;branch=z9hG4bK13
From:
To:
CSeq: 101INVITE
v=0
o=CiscoSystemsSIP-GW-UserAgent IN IP
s=SIP Call
c=IN IP
t=0 0
m=audio RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
m=video RTP/AVP 112
b=TIAS:
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.

24. Debugs for SIP-SIP Video Contd …
Received:
ACK SIP/2.0 \
Via: SIP/2.0/TCP ;branch=z9hG4bK1384b1
From:
To: Date: Tue, 17 Jul :12:57 GMT
CSeq: 101 ACK
v=0
o=CiscoSystemsCCM-SIP IN IP
s=SIP Call
c=IN IP
t=0 0
m=audio RTP/AVP 0
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=ptime:20
m=video RTP/AVP 112
b=TIAS:
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1
Sent: ACK SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK48CB1
From:
To:
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent IN IP
s=SIP Call
c=IN IP
t=0 0
m=audio RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
m=video RTP/AVP 112
b=TIAS:
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1

25. Debugs for a basic DO-EO video call
Received:
INVITE SIP/2.0
Via: SIP/2.0/TCP ;branch=z9hG4bK
From:
To:
Call-ID:
Supported: timer,replaces
Min-SE: 1800
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Call-Info: <sip: :5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Session-Expires: 1800
Contact:
Max-Forwards: 69
Content-Length: 0
Sent:
INVITE SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK18D28
From:
To:
Call-ID:
CSeq: 101 INVITE
Contact:
Content-Length: 551
v=0
o=CiscoSystemsSIP-GW-UserAgent IN IP
s=SIP Call
c=IN IP
t=0 0
m=audio RTP/AVP
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 RTP/AVP 112
a=rtpmap:112 H264/90000
a=fmtp:112 profile-level-id=ABCDEF;sprop-parameter-sets=Z00AKAoWVAPAEPI=,aGFLjyA=;packetization-mode=1

26. Debugs video-ipipgw1#sh voip rtp connections
SDP info. in incoming 183/200OK
v=0
o=CiscoSystemsCCM-GW-UserAgent IN IP s=SIP Call c=IN IP t=0 0 m=audio RTP/AVP c=IN IP a=rtpmap::96 mpeg4-generic/ a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:19 CN/8000
m=video RTP/AVP 112
c=IN IP
b=TIAS:
a=rtpmap:112 H264/90000
SDP info. in outgoing 183/200OK
v=0
o=CiscoSystemsCCM-GW-UserAgent IN IP s=SIP Call c=IN IP t=0 0 m=audio RTP/AVP c=IN IP a=rtpmap::96 mpeg4-generic/ a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:19 CN/8000
m=video RTP/AVP 112
c=IN IP
b=TIAS:
a=rtpmap:112 H264/90000
video-ipipgw1#sh voip rtp connections
No active connections found

27. 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