Hello everyone, Pastebin here: http://pastebin.freeswitch.org/11525 I've got my pjsip profile configured for G722 only: <param name="codec-prefs" value="G722"/> Yet whenever I send calls using that profile it (mysteriously) indicates support for PCMU in the INVITE. The pastebin includes both the INVITE and "sofia status profile pjsip" to show that only G722 has been enabled. Where is PCMU coming from? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
can you do another trace to show the inbound invite too? On Wed, Dec 16, 2009 at 1:13 PM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Hello everyone,
Pastebin here:
http://pastebin.freeswitch.org/11525
I've got my pjsip profile configured for G722 only:
<param name="codec-prefs" value="G722"/>
Yet whenever I send calls using that profile it (mysteriously) indicates support for PCMU in the INVITE. The pastebin includes both the INVITE and "sofia status profile pjsip" to show that only G722 has been enabled. Where is PCMU coming from?
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Sure... The call comes up as PCMU: INVITE sip:5888@10.70.0.99 SIP/2.0 Call-ID: 80ea31a017f6de1d53e4a9c52f00 CSeq: 1 INVITE From: sip:9413122830@smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 Record-Route: <sip:10.70.0.65:5060;lr>,<sip:10.70.0.69;lr;transport=tcp> To: "5888" <sip:5888@10.70.0.99> Via: SIP/2.0/UDP 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP 10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 Content-Length: 206 Content-Type: application/sdp Contact: <sip:9413122830@10.70.0.69;transport=tcp> Max-Forwards: 70 User-Agent: Avaya CM/R015x.02.0.947.3 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: timer,replaces,join,histinfo,100rel Alert-Info: <cid:external@10.70.0.99>;avaya-cm-alert-type=external Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: sip:9413122830@smh.sip.local P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" History-Info: <sip:5888@10.70.0.99>;index=1,"5888" <sip:5888@10.70.0.99>;index=1.1 v=0 o=- 1 1 IN IP4 10.70.0.69 s=- c=IN IP4 10.70.0.22 b=AS:64 t=0 0 m=audio 2176 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 We don't support G729 so this call comes up as PCMU when we answer and then that codec is first in the codec list... On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do another trace to show the inbound invite too?
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
yah so the codec chosen by the inbound leg is always offered in the outbound sdp to try and prevent transcoding. if you set {absolute_codec_string=G722} in the bridge string you will bypass this feature. On Wed, Dec 16, 2009 at 1:41 PM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Sure... The call comes up as PCMU:
INVITE sip:5888@10.70.0.99 <sip%3A5888@10.70.0.99> SIP/2.0 Call-ID: 80ea31a017f6de1d53e4a9c52f00 CSeq: 1 INVITE From: sip:9413122830@smh.sip.local;tag=80ea31a017f6de1d43e4a9c52f00 Record-Route: <sip:10.70.0.65:5060;lr>,<sip:10.70.0.69;lr;transport=tcp> To: "5888" <sip:5888@10.70.0.99 <sip%3A5888@10.70.0.99>> Via: SIP/2.0/UDP 10.70.0.65:5060;branch=z9hG4bK838383030303565656105e9.0,SIP/2.0/TCP
10.70.0.69;psrrposn=2;received=10.70.0.69;branch=z9hG4bK80ea31a017f6de1d63e4a9c52f00 Content-Length: 206 Content-Type: application/sdp Contact: <sip:9413122830@10.70.0.69 <sip%3A9413122830@10.70.0.69> ;transport=tcp> Max-Forwards: 70 User-Agent: Avaya CM/R015x.02.0.947.3 Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH Supported: timer,replaces,join,histinfo,100rel Alert-Info: <cid:external@10.70.0.99 <cid%3Aexternal@10.70.0.99>
;avaya-cm-alert-type=external Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: sip:9413122830@smh.sip.local P-Charging-Vector: icid-value="AAS:13283-a031ea801def6179c4a3ed3f52" History-Info: <sip:5888@10.70.0.99 <sip%3A5888@10.70.0.99> ;index=1,"5888" <sip:5888@10.70.0.99 <sip%3A5888@10.70.0.99>>;index=1.1
v=0 o=- 1 1 IN IP4 10.70.0.69 s=- c=IN IP4 10.70.0.22 b=AS:64 t=0 0 m=audio 2176 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000
We don't support G729 so this call comes up as PCMU when we answer and then that codec is first in the codec list...
On Wed, Dec 16, 2009 at 2:26 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
can you do another trace to show the inbound invite too?
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, As always, thanks. I thought that might be it but I wanted to make sure. Thanks again! On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
yah so the codec chosen by the inbound leg is always offered in the outbound sdp to try and prevent transcoding. if you set {absolute_codec_string=G722} in the bridge string you will bypass this feature.
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
np On Wed, Dec 16, 2009 at 1:59 PM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Anthony,
As always, thanks. I thought that might be it but I wanted to make sure.
Thanks again!
On Wed, Dec 16, 2009 at 2:48 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
yah so the codec chosen by the inbound leg is always offered in the outbound sdp to try and prevent transcoding. if you set {absolute_codec_string=G722} in the bridge string you will bypass this feature.
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Try setting "absolute_codec_string" in the dialplan prior to the bridge: <action application="export" data="nolocal:absolute_codec_string=G722"/> Let us know if that does the trick. -MC On Wed, Dec 16, 2009 at 11:13 AM, Kristian Kielhofner < kristian.kielhofner@gmail.com> wrote:
Hello everyone,
Pastebin here:
http://pastebin.freeswitch.org/11525
I've got my pjsip profile configured for G722 only:
<param name="codec-prefs" value="G722"/>
Yet whenever I send calls using that profile it (mysteriously) indicates support for PCMU in the INVITE. The pastebin includes both the INVITE and "sofia status profile pjsip" to show that only G722 has been enabled. Where is PCMU coming from?
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
participants (3)
-
Anthony Minessale
-
Kristian Kielhofner
-
Michael Collins