[Freeswitch-users] Try to get rid of Speex codec on invite

[Steven Ayre]

Just to be clear in case you're new to SDP...

The m=audio line gives a list of codecs identified by by number. Many of
them are static numbers meaning they are assigned to particular codecs. The
list is at http://www.iana.org/assignments/rtp-parameters 0=PCMU 8=PCMA etc.

Anything in the 96-127 range is dynamic and identified by name on a a=rtpmap
line.

For static numbers it's allowed to have a a=rtpmap line, but not required.
But some devices are broken and require it anyway. verbose=sdp=true puts
a=rtpmap for all codecs, which will work on all devices but at the risk of
possibly exceeding the MTU.

-Steve





On 11 January 2011 22:54, Brian West <brian at freeswitch.org> wrote:

> You are making the wrong conclusion.  The SDP is being verbose.  That SDP
> is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN
>
> What you need to set is the variable verbose_sdp=true so that we do a
> complete filled out SDP because the device you're speaking to is being
> intellectually challenged.  See the codec list in the m= line?
>
> /b
>
> On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
>
> > Hello,
> >
> > when our Freeswitch sends an INVITE to a phone, it only offers speex
> > codec for voice:
> > m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13
> > a=rtpmap:98 SPEEX/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
>
>
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