[Freeswitch-users] Code negotiation - all codec offered

[Peter P GMX]

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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