VoIP Mailing List Archives

[mrabbitt at chief-tech...]

We are experiencing an issue with our Cisco 9971 and 8945 phones where H264 video calls are connecting at 176x144 resolution instead of 640x480.  Soft clients can connect at higher resolutions and the 9971 can even receive video at a higher resolution (although it still sends 176x144).

I contacted one of the developers and he suggested the passthrough of SDP attributes is not working correctly.  Has anyone else experienced this problem?  We're running Asterisk 11.8.1.


Below are the video parts of the sip debug for one of the phones during a video call.  Should I be seeing the "a=imageattr" in the SIP OK message?






<--- SIP read from UDP:10.168.154.71:5060 --->
INVITE sip:7872@10.162.26.15 ([email]sip%3A7872@10.162.26.15[/email]);user=phone SIP/2.0
Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3
From: "Shawn Hughes" <sip:7871@10.162.26.15 ([email]sip%3A7871@10.162.26.15[/email])>;tag=20bbc0df35ef052672e68696-0b174da0
To: <sip:7872@10.162.26.15 ([email]sip%3A7872@10.162.26.15[/email])>
Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71 (20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71)
Max-Forwards: 70
Date: Fri, 28 Mar 2014 13:51:41 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP8945/9.4.1
Contact: <sip:7871@10.168.154.71:5060;transport=udp>;video
Authorization: Digest username="7871",realm="asterisk",uri="sip:7872@10.162.26.15 ([email]sip%3A7872@10.162.26.15[/email]);user=phone",response="f51a7522b01c90b81509d2274e9b69bb",nonce="5b43e5a6",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
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-6.0.2,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 685
Content-Type: application/sdp
Content-Disposition: session;handling=optional


v=0
o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71
s=SIP Call
t=0 0
m=audio 10032 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 10.168.154.71
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 10034 RTP/AVP 97
c=IN IP4 10.168.154.71
b=TIAS:2000000
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 [x=640,y=480] [x=640,y=360] [x=352,y=288] [x=176,y=144] recv [x=640,y=480]
a=sendrecv
<------------->
--- (19 headers 24 lines) ---
Sending to 10.168.154.71:5060 (no NAT)
Using INVITE request as basis request - 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71 (20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71)
Found peer '7871' for '7871' from 10.168.154.71:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 9
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Found RTP video format 97
Found video description format H264 for ID 97
Capabilities: us - (gsm|ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|g729|ilbc|g722|slin16)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g722|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.168.154.71:10032
Peer video RTP is at port 10.168.154.71:10034
Looking for 7872 in from-internal (domain 10.162.26.15)
list_route: hop: <sip:7871@10.168.154.71:5060;transport=udp>







SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3;received=10.168.154.71
From: "Shawn Hughes" <sip:7871@10.162.26.15 ([email]sip%3A7871@10.162.26.15[/email])>;tag=20bbc0df35ef052672e68696-0b174da0
To: <sip:7872@10.162.26.15 ([email]sip%3A7872@10.162.26.15[/email])>;tag=as1c2f9ae5
Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71 (20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71)
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.8.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:7872@10.162.26.15:5060>
Content-Type: application/sdp
Content-Length: 467


v=0
o=root 283568327 283568327 IN IP4 10.162.26.15
s=Asterisk PBX 11.8.1
c=IN IP4 10.162.26.15
b=CT:36000000
t=0 0
m=audio 13434 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 15496 RTP/AVP 97
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428014;max-mbps=36000;max-fs=1200;packetization-mode=0;level-asymmetry-allowed=1
a=sendrecv

[jcolp at digium.com]

Matt Rabbitt wrote:

[mrabbitt at chief-tech...]

What would need to be changed in the source code to accommodate this?  Can the imageattr attribute be hard coded into h264_format_attr_sdp_generate() in res_format_attr_h264.c?


On Mon, Mar 31, 2014 at 9:07 AM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:

[jcolp at digium.com]

Matt Rabbitt wrote: