[Freeswitch-users] Code negotiation - all codec offered

[Steven Ayre]

What do you have defined in the SIP profile for the following parameters?

outbound-codec-prefs
codec-prefs

You should have either outbound-codec-prefs or codec-prefs set to the
variable you set in vars.xml, you can also put the codec choices into
the SIP profile directly if you wish.

Also, just to check it's correct, what do you have defined for the
codecs in the vars.xml file?

-Steve


On 21 April 2010 11:23, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello
>
> since upgrading from 15648 to "FreeSWITCH Version 1.0.head (svn-17188)"
> we see that - despite only allowing PMCA and PCMU codecs in vars.xml -
> Freeswitch offers all available codec to the called phone.
>
> Any idea how to solve this?
>
>
> Best regards
> Peter
>
> See Invite  example
>
> U 10.xx.xx.1:5060 -> 10.xx.xx.172:2048
> INVITE sip:31 at 10.xx.xx.172:2048;line=hxbudrul SIP/2.0.
> Via: SIP/2.0/UDP 10.xx.xx.1;rport;branch=z9hG4bKZD66c84339SHH.
> Max-Forwards: 70.
> From: "069xxxxxxxx" <sip:069xxxxxxxx at 10.xx.xx.1>;tag=5evr6508K9S3K.
> To: <sip:31 at 10.xx.xx.172:2048;line=hxbudrul>.
> Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5.
> CSeq: 129803060 INVITE.
> Contact: <sip:mod_sofia at 10.xx.xx.1:5060>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-17188.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 920.
> X-FS-Support: update_display.
> Remote-Party-ID: "069xxxxxxxx"
> <sip:069xxxxxxxx at 10.xx.xx.1>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1271830560 1271830561 IN IP4 10.xx.xx.1.
> s=FreeSWITCH.
> c=IN IP4 10.xx.xx.1.
> t=0 0.
> m=audio 12232 RTP/AVP 115 96 99 18 4 7 124 8 6 123 0 10 98 122 9 121 3
> 101 13.
> a=rtpmap:115 G7221/32000.
> a=fmtp:115 bitrate=48000.
> a=rtpmap:96 AMR/8000.
> a=fmtp:96 octet-align=0.
> a=rtpmap:99 SPEEX/32000.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:7 LPC/8000.
> a=rtpmap:124 G726-16/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:6 DVI4/16000.
> a=rtpmap:123 G726-24/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:10 L16/22050.
> a=rtpmap:98 iLBC/8000.
> a=fmtp:98 mode
> #
>
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