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@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@10.xx.xx.1>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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 #
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@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@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@10.xx.xx.1>;tag=5evr6508K9S3K. To: <sip:31@10.xx.xx.172:2048;line=hxbudrul>. Call-ID: ad359ddd-c7cc-122d-3683-001517c965a5. CSeq: 129803060 INVITE. Contact: <sip:mod_sofia@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@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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codec-prefs will define both. inbound-codec-prefs outbound-codec-prefs were added. /b On Apr 21, 2010, at 5:23 AM, Peter P GMX 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
participants (3)
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Brian West
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Peter P GMX
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Steven Ayre