DanisHack / rtmplite

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Interoperate with Asterisk for Audio/Video #11

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
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

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