[Freeswitch-users] Outbound SIP invites without codec list

[Liam Farr]

Hi,



I set <action application="export" data="verbose_sdp=true" /> in my dial
plan and this fixed it.



My provider is rather strict about going through a testing / certification
process for any “new” hardware that they connect to their network, and
wanted to see the full codec list in the SIP invite, (even though as you say
it’s not required under the RFC’s).





Thanks



Liam



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre
*Sent:* Wednesday, 25 May 2011 6:22 p.m.
*To:* FreeSWITCH Users Help
*Subject:* Re: [Freeswitch-users] Outbound SIP invites without codec list



"m=audio 18646 RTP/AVP 0 8 101 13"

That is the codec list.
  0 = PCMU (G711 u-law)
  8 = PCMA (G711 a-law)
  101 = DTMF RFC2833
  13 = Comfort Noise

Numbers 96-127 are dynamic. Anything else is static and defined by IANA (
http://www.iana.org/assignments/rtp-parameters).

A dynamic payload type needs a rtpmap line to map the number to the codec
name. That is not required for static payload types, because they have a
preset rtpmap.

FreeSWITCH doesn't add the a=rtpmap lines for static types by default. The
advantage of doing so is it makes your packets smaller - less bandwidth
usage, and less likely to get a fragmented packet.

Some (very few) devices are broken though and require the rtpmap lines for
all codecs. These have broken SIP implementations because the RFC states the
rtpmap is optional for static types. There is a compatibility option if
you're using one of those and really do require it, but changes are you
don't: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

-Steve


On 25 May 2011 01:56, Liam Farr <liam at intersys.co.nz> 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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