Reported by Borja SIXTO <borja.sixto@i6net.com> on May 14, 2010
Hi,
I am Borja SIXTO from i6net.
We made working your Application (SipRTMP) with an Asterisk (audio and
video : using a transcoder in the Asterisk).
Are you interested by the changes done ?
Regards,
Borja
---
> Hi Borja,
>
> Thanks for the offer to send the changes.
> It will help a lot to the project if you can send your changes and any
> other lessons learned.
>
> Thanks.
---
Nice to get your response.
I do two modifications :
1/ enable Speex to be able to pass full audio calls with the Asterisk
(with Asterisk 1.6 it works fine, with a warning but the audio is ok).
2/ I had developp a video transcoder for the video h263sorenon
(integrated in the Asterisk 1.6), now I can call sip phone (h263, h263
and h264) and 3G video phones. For the video, the GW need to use 2
different netstreams to get a smooth audio/video.
It will be great to get the DTMF, but I don't know how to add the in
the SIP/sdp and how to send them from the P2P-sip API.
This is a prototype to make our native RTMP channel in the Asterisk.
Regards,
Borja
Original issue reported on code.google.com by voiprese...@gmail.com on 3 Feb 2011 at 11:55
Original issue reported on code.google.com by
voiprese...@gmail.comon 3 Feb 2011 at 11:55Attachments: