Created by: Denis Moiseenko on 30-09-2013 03:59:43 AM Hello! Could somebody please help me! I try to integrate third-party SIP Device (Mobotix T24), CUCM and cisco ip phone 8945. I've registered Mobotix T24 on CUCM as third-party SIP device (advanced) and also I've registered cisco ip phone 8945 as SIP device. When I call from any endpoint (Mobotix T24 or cisco phone 8945) I can establish just audio call between t24 and phone 8945. When I check call status on cisco phone I see it's receiving video streem with resolution 352x288 but the video never ever displays on cisco phone 8945. I checked logs from CUCM and I see that third party-device advertise video with rtp payload type - 103 98 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.102.13:5060;branch=z9hG4bK14f4065f789 From: <sip:email@example.com>;tag=3824~26574cc5-a750-4483-972d-d6dd16fee221-31947893 To: <sip:firstname.lastname@example.org>;tag=1887240230 Call-ID: email@example.com CSeq: 101 INVITE Contact: <sip:firstname.lastname@example.org:5060> Content-Type: application/sdp User-Agent: Linphone/3.0.0 MX Video (eXosip2/3.1.0) Content-Length: 466 v=0 o=3003 123456 654321 IN IP4 192.168.51.49 s=A conversation c=IN IP4 192.168.51.49 t=0 0 m=audio 7078 RTP/AVP 0 8 3 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 103 98 a=sendonly a=rtpmap:103 H264/90000 a=fmtp:103 profile-level-id=42800d;packetization-mode=1;level-asymmetry-allowed=1 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 but cisco ip phone advertise video with rtp payload type - 126 97 INVITE sip:email@example.com;user=phone SIP/2.0 Via: SIP/2.0/TCP 192.168.151.33:53225;branch=z9hG4bK60dae47a From: "5007" <sip:firstname.lastname@example.org>;tag=8478acedb7e500ae04185099-2be5d845 To: <sip:email@example.com> Call-ID: firstname.lastname@example.org Max-Forwards: 70 Date: Fri, 27 Sep 2013 10:40:56 GMT CSeq: 101 INVITE User-Agent: Cisco-CP8945/9.3.1 Contact: <sip:email@example.com:53225;transport=tcp>;video Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Remote-Party-ID: "5007" <sip:firstname.lastname@example.org>;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1 Allow-Events: kpml,dialog Content-Length: 948 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 23994 0 IN IP4 192.168.151.33 s=SIP Call t=0 0 m=audio 16388 RTP/AVP 0 8 18 102 9 116 101 c=IN IP4 192.168.151.33 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:102 L16/16000 a=rtpmap:9 G722/8000 a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 16390 RTP/AVP 126 97 c=IN IP4 192.168.151.33 b=TIAS:2000000 a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=428014;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300 a=imageattr:126 send recv a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200 a=imageattr:97 send recv a=sendrecv I have no way to change rtp payload type on the third-party device. Is it possible to change payload type on the side of CUCMusing SIP Normalization Scrip? I'm not familiar with SIP Normalization Scrip. May be somebody can help me or the references where I can find the answer. Also logs from CUCM in attachment. Thank you in advance!
Subject: RE: How to change rtp payload type using SIP Normalization Scrip in CUCM? Replied by: Mark Stover on 02-10-2013 10:30:45 AM Hi Denis,
You can certainly use Normalization to change the payload types on the SDP m-line. I can't guarantee that the endpoints will accept things, but you can change them as they cross Unified CM.
There are specific API calls that allow you to grab specific SDP lines (like the m= line) and modify them.
Subject: RE: How to change rtp payload type using SIP Normalization Scrip in CUCM? Replied by: Denis Moiseenko on 02-10-2013 01:14:22 PM Hi Mark,
Thank you very much for your response! I very appreciate it! I'm not familiar with language called "Lua" which is used for writing Normalization scripts. I tried to create one but it doesn't work. Maybe you can give me a link where I can find example how to change SDP line with payload type?
Thank you very much! -Denis
Subject: RE: How to change rtp payload type using SIP Normalization Scrip in CUCM? Replied by: Mark Stover on 08-10-2013 12:44:53 PM Sorry for delayed reply.
There are some examples provided in the Normalization Developer guide. Scroll down to the SIP Normalization and Transparency section of the SIP developer page:
If you have access to CiscoLive 365, I have a session that covers the basics of getting started there. Look for BRKCOL-2455. The session will (hopefully) be updated, expanded, and redelivered at Cisco Live in Milan and San Francisco this year.
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