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@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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