[Freeswitch-users] Codec Negotiation: Codec telephone-event:101:8000:20
Anthony Minessale
anthony.minessale at gmail.com
Wed Jan 6 17:51:12 PST 2010
when they only return 1, that means that is what they want you to use.
The call should establish in that case with g726-32
On Wed, Jan 6, 2010 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:
> NO the codec map includes 2 aka g726-32 which you don't have to list in the
> map if its defined in the standard... what I need is a pcap of the whole
> process.
>
> /b
>
> On Jan 6, 2010, at 6:43 PM, Mark Campbell-Smith wrote:
>
> > Thanks guys for your response. I'll have a read through the links sent
> to me.
> >
> > On the INVITE my FS box sends to Phonzo (the SIP provider) the codecs
> > it supports are G726-32, PCMU, PCMA.
> >
> > v=0
> > o=FreeSWITCH 1262742568 1262742569 IN IP4 xxx.xxx.xxx.xx
> > s=FreeSWITCH
> > c=IN IP4 xxx.xxx.xxx.xxx
> > t=0 0
> > m=audio 31566 RTP/AVP 2 0 8 101 13
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=rtpmap:13 CN/8000
> > a=ptime:20
> >
> > However, Phonzo responds with the following in the Session Progress
> message:
> >
> > v=0
> > o=Sippy 158698636 1 IN IP4 80.232.37.178
> > s=-
> > t=0 0
> > m=audio 61812 RTP/AVP 2 101 13
> > c=IN IP4 213.50.91.3
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> >
> > Does that make sense? The codec rtpmap only included DTMF ...
>
>
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--
Anthony Minessale II
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