[Freeswitch-users] Help debugging INVITE with provider

[Brian West]

Well looking at the invite from your provider:

v=0
o=FrancoisLegal 1275590034 1275590034 IN IP4 80.239.235.114
s=SIP Call
c=IN IP4 80.239.235.114
t=0 0
m=audio 11198 RTP/AVP 2 3 9 8 0 101
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 11340 RTP/AVP 31 34 115 121 99
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:115 H263-1998/90000
a=rtpmap:121 <-- THIS LINE
a=rtpmap:99 <-- AND THIS LINE
a=ptime:20

RFC4566 section 6:

RTP profiles that specify the use of dynamic payload types MUST define the set of valid encoding names and/or a means to register encoding names if that profile is to be used with SDP.

We are left guessing what 99 and 121 are which is invalid.

/b


On Jun 4, 2010, at 2:57 AM, François Legal wrote:

> This is what I did (except for the last trace) : MyProviderIp is the IP FS tries to contact to reach SomeAccount at provider.com, and MyFSIpHere is the IP FS external profile is bound to.
> 
> Maybe this lack some information : the "dialed number" is originate sofia/external/SomeAccount at provider.com
> 
> which should at the end come back to my FS external profile through the gateway 12voip.com included in the external profile.
> 
>  
> François
>