[Asterisk-video] Branches Videocaps

[Klaus Darilion]

hi!

Unfortunately I have not tried the videocaps branch yet. I once had BW 
problems too and I solved it with this patch: 
http://bugs.digium.com/view.php?id=12134

This works for xlite/eyebeam.

regards
klaus

Valerio Puglia schrieb:
> yes it's work on 1.6 beta 7.1 with the same problem i describe below:
> 
> i try to use videcaps for settings a videocall bandwith and other option..
> 
> I have made a mistake:
> the kapanga works but the video is terrible.... in incoming calls but is 
> perfect on videcall out..
> the sdp that kapanga send to outgoing calls to asterisk is
> 
> o=9001 1208967526 1208968601 IN IP4 10.0.2.182
> s=Kapanga [1208967526]
> c=IN IP4 10.0.2.182
> t=0 0
> m=audio 5130 RTP/AVP 0 8 101
> a=rtpmap:0 pcmu/8000
> a=sendrecv
> a=silenceSupp:off - - -
> a=rtcp:5131
> a=maxptime:20
> a=ptime:20
> a=rtpmap:8 pcma/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15,36
> m=video 5132 RTP/AVP 98 105
> a=rtpmap:98 theora/90000
> a=fmtp:98 sampling=YCbCr-4:2:0; width=176, height=144;
> b=AS:61
> a=sendrecv
> a=rtcp:5133
> a=rtpmap:105 h263-1998/90000
> a=fmtp:105 profile=0; level=10
> 
> on videocall out kapanga set the bandwith information with b=AS:xx
> and video works very well
> but on sip.conf is possible to set only b=CT:XXX with the option 
> maxcallbitrate=xxx
> is possible to set maxvideobitrate (b=AS:xx)?
> 
> because i think it is caused by that (the video bitrate too high)
> 
> 
> 
> 
> Klaus Darilion ha scritto:
>> does it work with plain Asterisk (not videocaps branch)?
>>
>> regards
>> klaus
>>
>> Valerio Puglia schrieb:
>>   
>>> I try to use this branches with fontventa application
>>>
>>> the version is svn 114578
>>>
>>> when the call is redirect on siphone eyeBeam 1.5 or kapanga the video 
>>> sent by cellular doesn't start...on x lite give this erorr "SIP/2.0 415 
>>> Unsupported Media Type"
>>>
>>> the sdp of invite is
>>>
>>> v=0
>>> o=root xxxx
>>> s=Asterisk PBX SVN-oej-videocaps-r114506M-/trunk
>>> c=IN IP4 x.x.x.x
>>> b=CT:6000
>>> t=0 0
>>> m=audio 11428 RTP/AVP 0 8 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>> m=video 10488 RTP/AVP 34 98
>>> a=rtpmap:34 H263/90000
>>> a=fmtp:34;CIF=1;QCIF=1;maxbr=3840
>>> a=rtpmap:98 h263-1998/90000
>>> a=fmtp:98;CIF=1;QCIF=1;maxbr=3840
>>> a=sendrecv
>>>
>>>
>>>
>>> CSeq: 102 INVITE
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
>>> SUBSCRIBE, INFO
>>> Content-Type: application/sdp
>>> User-Agent: eyeBeam release 1100l stamp 46320
>>> Content-Length: 250
>>>
>>> v=0
>>> o=- 1 2 IN IP4 10.0.2.5
>>> s=CounterPath eyeBeam 1.5
>>> c=IN IP4 62.94.144.245
>>> t=0 0
>>> m=audio 37208 RTP/AVP 0 8 101
>>> a=fmtp:101 0-15
>>> a=rtpmap:101 telephone-event/8000
>>> a=sendrecv
>>> m=video 44800 RTP/AVP 98
>>> a=rtpmap:98 H263-1998/90000
>>> a=sendrecv
>>>
>>> i try to change the parameter of video codec on sip.conf to
>>>
>>> h263=cif=30 qcif=30 maxbr=1024
>>> h263p=cif=30 qcif=30 maxbr=1024
>>>
>>> but in sdp nothing has changed "a=fmtp:98;CIF=1;QCIF=1;maxbr=3840"
>>>
>>> on outgoing call the video on sipphone is ok
>>>
>>> any idea?
>>>
>>>     
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>   
> 
>