[Freeswitch-users] Outbound SIP invites without codec list

[Steven Ayre]

"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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