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"
does it work with plain Asterisk (not videocaps branch)?
regards
klaus
Valerio Puglia schrieb:
Quote:
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"
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
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:
Quote:
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--
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
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?
>>
>>
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