[Speex-dev] app_conference(Asterisk) with Speex
Jean-Marc.Valin at USherbrooke.ca
Tue Jan 31 14:24:30 PST 2006
Just curious, how does Asterisk pack Speex frames in a packet. AFAIK,
Linphone just sends raw packets, as specified in the RTP draft.
Le mardi 31 janvier 2006 à 10:43 -0500, Steve Kann a écrit :
> jonathan blais wrote:
> > I'm using Linphone. I tested with Asterisk and Speex only, I created
> > a channel with echo and it worked. It seems to have problem when
> > using app_conference.
> If you just use app_echo, then asterisk won't be trying to decode your
> frames; it will just be sending them back to you. Therefore, if your
> client is using an incompatible packing of the frames, it won't
> matter, as long as your client can also understand it.
> In order to determine if your client is using a compatible packing,
> you'll need to have asterisk do something which requires asterisk to
> decode the frames: Like bridge the call to a channel using another
> codec, or record the call to a non-speex format (i.e. write to a PCM
> WAV file), etc.
> > Jonathan
> > 2006/1/31, Steve Kann <stevek at stevek.com>:
> > jonathan blais wrote:
> > > Hi,
> > >
> > > Does anyone ever used Speex with app_conference in
> > Asterisk ? I'm
> > > having a hard time to figure why I always get this error
> > "warning:
> > > Invalid mode encountered: corrupted stream?".
> > Yes, we do this all the time. Where are your speex frames
> > coming from?
> > There's some SIP phones out there which seem to use an
> > incompatible
> > speex encoding, I seem to recall.
> > -SteveK
> Speex-dev mailing list
> Speex-dev at xiph.org
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