<div dir="ltr"><div class="gmail_quote"><div dir="ltr">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).
<div><br></div><div>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.</div><div>
<br></div><div>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?</div><div><br></div><div><br></div><div><br></div>
<div><div><--- SIP read from UDP:<a href="http://10.168.154.71:5060" target="_blank">10.168.154.71:5060</a> ---></div><div>INVITE <a href="mailto:sip%3A7872@10.162.26.15" target="_blank">sip:7872@10.162.26.15</a>;user=phone SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3</div>
<div>From: "Shawn Hughes" <<a href="mailto:sip%3A7871@10.162.26.15" target="_blank">sip:7871@10.162.26.15</a>>;tag=20bbc0df35ef052672e68696-0b174da0</div><div>To: <<a href="mailto:sip%3A7872@10.162.26.15" target="_blank">sip:7872@10.162.26.15</a>></div>
<div>Call-ID: <a href="mailto:20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71" target="_blank">20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71</a></div><div>Max-Forwards: 70</div><div>Date: Fri, 28 Mar 2014 13:51:41 GMT</div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Cisco-CP8945/9.4.1</div><div>Contact: <sip:7871@10.168.154.71:5060;transport=udp>;video</div><div>Authorization: Digest username="7871",realm="asterisk",uri="<a href="mailto:sip%3A7872@10.162.26.15" target="_blank">sip:7872@10.162.26.15</a>;user=phone",response="f51a7522b01c90b81509d2274e9b69bb",nonce="5b43e5a6",algorithm=MD5</div>
<div>Expires: 180</div><div>Accept: application/sdp</div><div>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO</div><div>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</div>
<div>Allow-Events: kpml,dialog</div><div>Content-Length: 685</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session;handling=optional</div><div><br></div><div>v=0</div><div>o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71</div>
<div>s=SIP Call</div><div>t=0 0</div><div>m=audio 10032 RTP/AVP 0 8 18 102 9 116 101</div><div>c=IN IP4 10.168.154.71</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:18 G729/8000</div><div>
a=fmtp:18 annexb=no</div><div>a=rtpmap:102 L16/16000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:116 iLBC/8000</div><div>a=fmtp:116 mode=20</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div>
<div>a=sendrecv</div><div>m=video 10034 RTP/AVP 97</div><div>c=IN IP4 10.168.154.71</div><div>b=TIAS:2000000</div><div>a=rtpmap:97 H264/90000</div><div>a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200</div>
<div>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]</div><div>a=sendrecv</div><div><-------------></div><div>--- (19 headers 24 lines) ---</div><div>Sending to <a href="http://10.168.154.71:5060" target="_blank">10.168.154.71:5060</a> (no NAT)</div>
<div>Using INVITE request as basis request - <a href="mailto:20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71" target="_blank">20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71</a></div><div>Found peer '7871' for '7871' from <a href="http://10.168.154.71:5060" target="_blank">10.168.154.71:5060</a></div>
<div>Found RTP audio format 0</div><div>Found RTP audio format 8</div><div>Found RTP audio format 18</div><div>Found RTP audio format 102</div><div>Found RTP audio format 9</div><div>Found RTP audio format 116</div><div>
Found RTP audio format 101</div>
<div>Found audio description format PCMU for ID 0</div><div>Found audio description format PCMA for ID 8</div><div>Found audio description format G729 for ID 18</div><div>Found audio description format L16 for ID 102</div>
<div>Found audio description format G722 for ID 9</div><div>Found audio description format iLBC for ID 116</div><div>Found audio description format telephone-event for ID 101</div><div>Found RTP video format 97</div><div>
Found video description format H264 for ID 97</div><div>Capabilities: us - (gsm|ulaw|alaw|g722|h264), peer - audio=(ulaw|alaw|g729|ilbc|g722|slin16)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g722|h264)</div><div>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)</div><div>Peer audio RTP is at port <a href="http://10.168.154.71:10032" target="_blank">10.168.154.71:10032</a></div>
<div>Peer video RTP is at port <a href="http://10.168.154.71:10034" target="_blank">10.168.154.71:10034</a></div><div>Looking for 7872 in from-internal (domain 10.162.26.15)</div><div>list_route: hop: <sip:7871@10.168.154.71:5060;transport=udp></div>
</div><div><br></div><div><br></div><div><br></div><div><div>SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3;received=10.168.154.71</div><div>From: "Shawn Hughes" <<a href="mailto:sip%3A7871@10.162.26.15" target="_blank">sip:7871@10.162.26.15</a>>;tag=20bbc0df35ef052672e68696-0b174da0</div>
<div>To: <<a href="mailto:sip%3A7872@10.162.26.15" target="_blank">sip:7872@10.162.26.15</a>>;tag=as1c2f9ae5</div><div>Call-ID: <a href="mailto:20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71" target="_blank">20bbc0df-35ef000a-453db49e-67cd30f1@10.168.154.71</a></div>
<div>CSeq: 102 INVITE</div><div>Server: FPBX-2.11.0(11.8.1)</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Contact: <<a href="http://sip:7872@10.162.26.15:5060" target="_blank">sip:7872@10.162.26.15:5060</a>></div>
<div>Content-Type: application/sdp</div><div>Content-Length: 467</div><div><br></div><div>v=0</div><div>o=root 283568327 283568327 IN IP4 10.162.26.15</div><div>s=Asterisk PBX 11.8.1</div><div>c=IN IP4 10.162.26.15</div>
<div>
b=CT:36000000</div><div>t=0 0</div><div>m=audio 13434 RTP/AVP 0 8 9 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div>
<div>a=ptime:20</div><div>a=sendrecv</div><div>m=video 15496 RTP/AVP 97</div><div>a=rtpmap:97 H264/90000</div><div>a=fmtp:97 profile-level-id=428014;max-mbps=36000;max-fs=1200;packetization-mode=0;level-asymmetry-allowed=1</div>
<div>a=sendrecv</div></div></div></div>
</div>