asterisk users - Mar 2014 - Video calls using Cisco phones are 176x144(QCIF) and 15FPS both ways

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 at 10.162.26.15;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.168.154.71:5060;branch=z9hG4bK1182b2d3
From: "Shawn Hughes" <sip:7871 at
10.162.26.15
>;tag=20bbc0df35ef052672e68696-0b174da0
To: <sip:7872 at 10.162.26.15>
Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1 at 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 at 10.168.154.71:5060;transport=udp>;video
Authorization: Digest
username="7871",realm="asterisk",uri="
sip:7872 at 10.162.26.15
;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 at 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 at 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 at
10.162.26.15
>;tag=20bbc0df35ef052672e68696-0b174da0
To: <sip:7872 at 10.162.26.15>;tag=as1c2f9ae5
Call-ID: 20bbc0df-35ef000a-453db49e-67cd30f1 at 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 at 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
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Matt Rabbitt wrote:
> 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?
It looks as though the passthrough for "fmtp" is indeed working but as
the "imageattr" attribute is currently unsupported/not used/not passed
through it is probably causing your resolution problem.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org