[Freeswitch-users] static rtpmap entries are not shown in outbound invites

[Leon de Rooij]

Hi,

When I'm using late negotiation and in my dialplan do the following:

<action application="export" data="codec_string=${ep_codec_string}"/>
<action application="set" data="inherit_codec=true"/>

(I also tried with absolute_codec_string instead of codec_string)

Then the codecs (or more precisely said, the rtpmap lines) that are offered in the SDP of the A leg are not offered to the b-leg. Only the RTP payload numbers as defined by IANA are there in the media (m) header.

For example:


a-leg SDP in INVITE:

m=audio 16864 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.


b-leg SDP in INVITE:

m=audio 24148 RTP/AVP 18 101 13.
a=rtpmap:101 telephone-event/8000.


Is there a reason why FS removes the G729 rtpmap, even though I exported all into the codec_string chan var ?

Now, this would normally be fine because G729 being 18 is a static rtpmap, but I have one endpoint that has the behavior that if I send an invite with RTP payload 18 and omit the rtpmap being G729, then it responds with a 183 session progress containing rtpmap G.729 (notice the dot) - this is wrong as far as I can tell.

Apparently they respond with the correct G729 (without the dot) in their 183 sess progress if I include the rtpmap in my invite.

I believe I read somewhere there was an option settable (on a sip profile?) that enables to always send all payloads as defined in the media (m) sdp header as a=rtpmap lines, but I cannot find the option anymore (or perhaps I'm mistaken and saw it somewhere else ?)

Can someone help me with this ?

Thanks & kind regards,

Leon