[Asterisk-Dev] silencesupp header in SDP

[Steve Langstaff]

If Asterisk uses incoming voice frames as a timer to send it's own voice frames,
does this preclude asterisk servers ever sending audio to each other? Who would start? :)

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of Moises Silva
Sent: 17 January 2006 14:51
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] silencesupp header in SDP


Youre right, chan_sip.c is the right place. But AFAIK Asterisk does
not support silence suppression. If you remove that asterisk may stop
sending sound, because it seems it uses  the incoming voice frames
(including silence frames) as timer to send its own sound frames. 
Plase correct me if im wrong.

Regards

On 1/17/06, Atif Rasheed <atif at iphonica.com> wrote:
> Sorry, its not silencesupp attribute, its Annex-B indication which was
> not in 1.0.9 and now its there in 1.2.1 and probably here parsing is
> failing.
>
> Media Attribute ( a ): fmtp:18 annexb=no
>                  Media Attribute Fieldname: fmtp
>                  Media Attribute Value: 18 annexb=no
>
>
> Shoud I remove it from chan_sip.c, i.e. I just remove this block
>  if (codec == AST_FORMAT_G729A)
>                 /* Indicate that we don't support VAD (G.729 annex B) */
>                 ast_build_string(a_buf, a_size, "a=fmtp:%d
> annexb=no\r\n", rtp_code);
>
> and Annex-B indication will not be there in outgoing SDP messages ???
>
> Thank you
>
> Atif
>
> Atif Rasheed wrote:
>
> > hello all,
> >
> > How can I strip this header from outgoing SDP messages
> >
> > Media Attribute (a): silenceSupp:off - - - -
> >                Media Attribute Fieldname: silenceSupp
> >                Media Attribute Value: off - - - -
> >
> > probably it was not there in 1.0.9, but when I upgraded to 1.2.1 its
> > creating problems for some GW's .
> >
> > Thank you
> >
> > Atif
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