[Freeswitch-users] sdp header rewrite

[Gabriel Kuri]

I'm having an issue with the linksys spa devices when enabling inbound
proxy media mode (inbound-proxy-media=true) and late negotiation
(inbound-late-negotiation=true) in the sofia profile. The spa
immediately sends a BYE when the call is answered by the called party.
For whatever reason, it works fine between two linksys devices directly
connected to FS, but when the call goes out to the PSTN via the SIP
provider, the spa isn't happy and sends a BYE.

After comparing the raw SIP packets on the wire (tcpdump) and between
enabling/disabling proxy-media mode and late negotiation, the only
difference I notice is the port in the m= line of the SDP header.

According to the freeswitch log, the rtp port would be rewritten to
28044 in the sdp header of the SIP packet sent to the spa device. But on
the wire, the port is rewritten to 0, which I'm guessing is why the spa
isn't happy and sending a BYE.

Here's the excerpt from the freeswitch log showing FS rewriting the port
to 28044 for the packet going to the spa device.


[DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp()
sofia/internal/<phone_number_removed>@mydomain.net Patched SDP
---
v=0
o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
s=sip call
c=IN IP4 XX.XX.XX.XX
t=0 0
m=audio 24174 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

+++
v=0
o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
s=sip call
c=IN IP4 YY.YY.YY.YY
t=0 0
m=audio 28044 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20


However, the packet on the wire reveals FS rewriting the port to 0  ...

v=0.
o=01Nextone 7852943629956191733 8120394851828294756 IN IP4 YY.YY.YY.YY.
s=sip call.
c=IN IP4 YY.YY.YY.YY.
t=0 0.
m=audio 0 RTP/AVP 96 101.
a=rtpmap:96 G729/8000.
a=fmtp:96 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

Is this a bug or is there some other problem?

Thanks for the help,
Gabe