[Freeswitch-users] Outbound SIP invites without codec list
Spencer Thomason
spencer at 5ninesolutions.com
Wed May 25 08:43:33 MSD 2011
There is actually a codec list but those are static RTP payload types.
You are offering PCMU, PCMA, RFC2833 DTMF, and Comfort Noise
See:
http://www.iana.org/assignments/rtp-parameters
If you need the full SDP for even known static types set verbose_sdp=true in vars.xml.
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
Spencer
On May 24, 2011, at 7:56 PM, Liam Farr wrote:
> Hi,
>
> My outbound SIP invites are going out without a codec list? Could someone point me in the right direction on how to set / insert this?
>
>
> send 972 bytes to udp/[1.2.3.4]:5060 at 00:28:52.144565:
> ------------------------------------------------------------------------
> INVITE sip:0800000000 at 1.2.3.4:5060 SIP/2.0
> Via: SIP/2.0/UDP 9.8.7.6;rport;branch=z9hG4bK2QeQtFFa7Z6Hr
> Max-Forwards: 69
> From: "Liam - Test" <sip:12345678 at 9.8.7.6>;tag=ryFQea59e6Upe
> To: <sip:0800000000 at 1.2.3.4:5060>
> Call-ID: cec20463-0108-122f-689e-c99dbf5a2f43
> CSeq: 12805602 INVITE
> Contact: <sip:12345678 at 9.8.7.6:5060>
> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 189
> X-FS-Support: update_display
> Remote-Party-ID: "Liam - Test" <sip:12345678 at 9.8.7.6>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1306264686 1306264687 IN IP4 9.8.7.6
> s=FreeSWITCH
> c=IN IP4 9.8.7.6
> t=0 0
> m=audio 18646 RTP/AVP 0 8 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> ------------------------------------------------------------------------
>
>
> Thanks
>
> Liam
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