Number of codecs offerred in SDP
Test Setup: vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/> The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here. FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension. m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30 Thanks
<param name="inbound-late-negotiation" value="true"/> in SIP profile settings. On Wednesday 23 June 2010, Mark Campbell-Smith wrote:
Test Setup:
vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30
Thanks
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Mark, that's not normal. FS should offer PCMU/G729/GSM to 1020. Can you check your SIP Profile parameters ? Is it configured with: <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> What FS version are you running ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 23/06/2010 à 14:07, Mark Campbell-Smith a écrit :
Test Setup:
vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30
Thanks
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I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? Wouldn't fs only get involved when its media server is referred to? If the "other endpoint" will only accept G729, doesn't that mean you need to change that endpoint to also accept G711 or also license G729 in FS? On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30
Thanks
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-- Sent from my mobile device ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431 Email: tgraziano@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation I have this set in my internal profile: <param name="codec-prefs" value="$${global_codec_prefs}"/> and as stated before vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/> Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously. On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano <tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you need to change that endpoint to also accept G711 or also license G729 in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30
Thanks
_______________________________________________ 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
-- Sent from my mobile device
====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
_______________________________________________ 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
Check to see if you have mod_spandsp or mod_voipcodecs loaded. /b On Jun 23, 2010, at 7:43 AM, Mark Campbell-Smith wrote:
Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile: <param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously.
Mark, I confirm that, as I wrote that wiki page (the early negotiation part) :) Can you really confirm your FS version ? The parameter you showed is old. codec-prefs has been replaced in SIP profiles by: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$$ {outbound_codec_prefs}"/> David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile: <param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano <tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints? Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you need to change that endpoint to also accept G711 or also license G729 in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of codecs that are offered to an extension? I know I can change the order in the codec_prefs, but would prefer FS to offer all three codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30
Thanks
_______________________________________________ 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
-- Sent from my mobile device
====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgraziano@voice.myitdepartment.net Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.net Fax: 434.984.8427
Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
_______________________________________________ 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
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FS version and codecs are shown below, but my config file are probably quite old. But I guess they should still work? All codecs are loaded, and the call works if late negotiation is set on profile internal. As I wrote above: The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here. I forgot to write that Extension 1000 does not support GSM (I want to force transcoding). Is that why FS is filtering out GSM on the b-leg? freeswitch@internal> version FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500) freeswitch@internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_com_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex 25 total. On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Mark, I confirm that, as I wrote that wiki page (the early negotiation part) :) Can you really confirm your FS version ? The parameter you showed is old. codec-prefs has been replaced in SIP profiles by: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile: <param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano <tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you
need to change that endpoint to also accept G711 or also license G729
in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of
codecs that are offered to an extension? I know I can change the
order in the codec_prefs, but would prefer FS to offer all three
codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
Thanks
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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--
Sent from my mobile device
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano@voice.myitdepartment.net
Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk@voice.myitdepartment.net
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Hi All, I'm not really sure if I got a firm answer for this one. Is the only way to ensure that transcoding is performed is by using late-negotiation? Why aren't all my codecs sent in the INVITE message to the B-leg (extension 1020)? Thanks! On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
FS version and codecs are shown below, but my config file are probably quite old. But I guess they should still work?
All codecs are loaded, and the call works if late negotiation is set on profile internal. As I wrote above: The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
I forgot to write that Extension 1000 does not support GSM (I want to force transcoding). Is that why FS is filtering out GSM on the b-leg?
freeswitch@internal> version FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
freeswitch@internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_com_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex
25 total.
On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Mark, I confirm that, as I wrote that wiki page (the early negotiation part) :) Can you really confirm your FS version ? The parameter you showed is old. codec-prefs has been replaced in SIP profiles by: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile: <param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano <tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you
need to change that endpoint to also accept G711 or also license G729
in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of
codecs that are offered to an extension? I know I can change the
order in the codec_prefs, but would prefer FS to offer all three
codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
Thanks
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano@voice.myitdepartment.net
Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk@voice.myitdepartment.net
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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Please, retry with a genuine config (the default one would be a wise choice). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :
Hi All,
I'm not really sure if I got a firm answer for this one. Is the only way to ensure that transcoding is performed is by using late-negotiation? Why aren't all my codecs sent in the INVITE message to the B-leg (extension 1020)?
Thanks!
On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
FS version and codecs are shown below, but my config file are probably quite old. But I guess they should still work?
All codecs are loaded, and the call works if late negotiation is set on profile internal. As I wrote above: The call setup is extension 1000 calls extension 1020 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by FS as the A-leg codec 2. Extension 1020 only supports GSM codec. The call fails with Not Acceptable Here.
I forgot to write that Extension 1000 does not support GSM (I want to force transcoding). Is that why FS is filtering out GSM on the b- leg?
freeswitch@internal> version FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
freeswitch@internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_voipcodecs codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_voipcodecs codec,G.723.1 6.3k,mod_g723_1 codec,G.726 16k,mod_voipcodecs codec,G.726 16k (AAL2),mod_voipcodecs codec,G.726 24k,mod_voipcodecs codec,G.726 24k (AAL2),mod_voipcodecs codec,G.726 32k,mod_voipcodecs codec,G.726 32k (AAL2),mod_voipcodecs codec,G.726 40k,mod_voipcodecs codec,G.726 40k (AAL2),mod_voipcodecs codec,G.729,mod_com_g729 codec,GSM,mod_voipcodecs codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_voipcodecs codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Speex,mod_speex
25 total.
On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com
wrote: Mark, I confirm that, as I wrote that wiki page (the early negotiation part) :) Can you really confirm your FS version ? The parameter you showed is old. codec-prefs has been replaced in SIP profiles by: <param name="inbound-codec-prefs" value="$$ {global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$$ {outbound_codec_prefs}"/> David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early negotiation default): http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile: <param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki page, I see the following sentence, which I interpret that GSM should still be sent: When FS calls leg B, the list of codecs in outbound-codec-prefs for the SIP profile is reorganized by pushing the codec negotiated above for leg A at the top . If B does not accept any of the codecs, the calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano <tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you
need to change that endpoint to also accept G711 or also license G729
in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of
codecs that are offered to an extension? I know I can change the
order in the codec_prefs, but would prefer FS to offer all three
codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
Thanks
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Sent from my mobile device
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgraziano@voice.myitdepartment.net
Fax: 434.984.8431
Email: tgraziano@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk@voice.myitdepartment.net
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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Updating the configuration did not help. I'm not telling FS to use any other codecs other than what I have specified below, so I'm not sure what I have done wrong. Below is the full trace: ------------------------------------------------------------------------ INVITE sip:1020@192.168.1.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae From: 1000 <sip:1000@192.168.1.120>;tag=2c7a518d12f9370eo0 To: <sip:1020@192.168.1.120> Call-ID: 316156b9-f6b413d2@192.168.1.121 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6-d9ad5b204ca2",uri="sip:1020@192.168.1.120",algorithm=MD5,response="c4a5e5cb60676366b8bcfb1329f3fc08",qop=auth,nc=00000001,cnonce="6a0220d0" Contact: 1000 <sip:1000@192.168.1.121:5060> Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 451 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 17789112 17789112 IN IP4 192.168.1.121 s=- c=IN IP4 192.168.1.121 t=0 0 m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv ------------------------------------------------------------------------ send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae From: 1000 <sip:1000@192.168.1.120>;tag=2c7a518d12f9370eo0 To: <sip:1020@192.168.1.120> Call-ID: 316156b9-f6b413d2@192.168.1.121 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 14-49-15 -0500 Content-Length: 0 ------------------------------------------------------------------------ 2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121 Rejected by acl "domains". Falling back to Digest auth. 2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1000@192.168.1.120 [77a3ab82-8447-11df-93a7-d9ad5b204ca2] 2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_NEW 2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000@192.168.1.120) State NEW 2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel sofia/internal/1000@192.168.1.120 entering state [received][100] 2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP: v=0 o=- 17789112 17789112 IN IP4 192.168.1.121 s=- c=IN IP4 192.168.1.121 t=0 0 m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[G729:18:8000:20] 2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[PCMU:0:8000:20] 2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[GSM:3:8000:20] 2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting codec PCMU@30i@8000h 2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec sofia/internal/1000@192.168.1.120 PCMU/8000 30 ms 240 samples 2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf send/recv payload to 101 2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451 (sofia/internal/1000@192.168.1.120) State Change CS_NEW -> CS_INIT 2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000@192.168.1.120 [BREAK] 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_INIT 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000@192.168.1.120) State INIT 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83 sofia/internal/1000@192.168.1.120 SOFIA INIT 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117 (sofia/internal/1000@192.168.1.120) State Change CS_INIT -> CS_ROUTING 2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000@192.168.1.120 [BREAK] 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000@192.168.1.120) State INIT going to sleep 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_ROUTING 2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474 (sofia/internal/1000@192.168.1.120) Callstate Change DOWN -> RINGING 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000@192.168.1.120) State ROUTING 2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333 (sofia/internal/1000@192.168.1.120) Callstate Change RINGING -> ACTIVE 2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140 sofia/internal/1000@192.168.1.120 SOFIA ROUTING 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000@192.168.1.120 Standard ROUTING 2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing 1000->1020 in context default Dialplan: sofia/internal/1000@192.168.1.120 parsing [default->Local_1000_1019] continue=false Dialplan: sofia/internal/1000@192.168.1.120 Regex (FAIL) [Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000@192.168.1.120 parsing [default->Mobile_1020s] continue=false Dialplan: sofia/internal/1000@192.168.1.120 Regex (PASS) [Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/ break=on-false Dialplan: sofia/internal/1000@192.168.1.120 Action set(dialed_extension=1020) Dialplan: sofia/internal/1000@192.168.1.120 Action export(codec_string=GSM) Dialplan: sofia/internal/1000@192.168.1.120 Action set(codec_string=GSM) Dialplan: sofia/internal/1000@192.168.1.120 Action bridge(user/${dialed_extension}@${domain}) Dialplan: sofia/internal/1000@192.168.1.120 Action set_user(1000@${domain}) Dialplan: sofia/internal/1000@192.168.1.120 Action answer() Dialplan: sofia/internal/1000@192.168.1.120 Action sleep(1000) Dialplan: sofia/internal/1000@192.168.1.120 Action system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount} ${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$ <action application=) 2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000@192.168.1.120) State Change CS_ROUTING -> CS_EXECUTE 2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000@192.168.1.120 [BREAK] 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000@192.168.1.120) State ROUTING going to sleep 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_EXECUTE 2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000@192.168.1.120) State EXECUTE 2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233 sofia/internal/1000@192.168.1.120 SOFIA EXECUTE 2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000@192.168.1.120 Standard EXECUTE EXECUTE sofia/internal/1000@192.168.1.120 set(dialed_extension=1020) 2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843 sofia/internal/1000@192.168.1.120 SET [dialed_extension]=[1020] EXECUTE sofia/internal/1000@192.168.1.120 bridge(user/1020@mydns.dyndns.org) 2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956 variable string 0 = [presence_id=1020@mydns.dyndns.org] 2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1020@192.168.1.123:5060 [77d7e3b6-8447-11df-93a8-d9ad5b204ca2] 2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883 (sofia/internal/sip:1020@192.168.1.123:5060) State Change CS_NEW -> CS_INIT 2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1020@192.168.1.123:5060 [BREAK] 2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1020@192.168.1.123:5060) Running State Change CS_INIT 2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1020@192.168.1.123:5060) State INIT 2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1020@192.168.1.123:5060 SOFIA INIT send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944: ------------------------------------------------------------------------ INVITE sip:1020@192.168.1.123:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ Max-Forwards: 69 From: "1000" <sip:1000@192.168.1.120>;tag=Z51Qve55SUHta To: <sip:1020@192.168.1.123:5060> Call-ID: 4f442091-feea-122d-448a-00e04c0312e9 CSeq: 132833080 INVITE Contact: <sip:mod_sofia@192.168.1.120:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 14-49-15 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display Remote-Party-ID: "1000" <sip:1000@192.168.1.120>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120 s=FreeSWITCH c=IN IP4 192.168.1.120 t=0 0 m=audio 27386 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30 It fails after this with Not Acceptable Here / INCOMPATIBLE_DESTINATION On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Please, retry with a genuine config (the default one would be a wise choice). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :
Hi All,
I'm not really sure if I got a firm answer for this one. Is the only way to ensure that transcoding is performed is by using late-negotiation? Why aren't all my codecs sent in the INVITE message to the B-leg (extension 1020)?
Thanks!
On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
FS version and codecs are shown below, but my config file are probably
quite old. But I guess they should still work?
All codecs are loaded, and the call works if late negotiation is set
on profile internal.
As I wrote above:
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
I forgot to write that Extension 1000 does not support GSM (I want to
force transcoding). Is that why FS is filtering out GSM on the b-leg?
freeswitch@internal> version
FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
freeswitch@internal> show codecs
type,name,ikey
codec,ADPCM (IMA),mod_voipcodecs
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_voipcodecs
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_voipcodecs
codec,G.726 16k (AAL2),mod_voipcodecs
codec,G.726 24k,mod_voipcodecs
codec,G.726 24k (AAL2),mod_voipcodecs
codec,G.726 32k,mod_voipcodecs
codec,G.726 32k (AAL2),mod_voipcodecs
codec,G.726 40k,mod_voipcodecs
codec,G.726 40k (AAL2),mod_voipcodecs
codec,G.729,mod_com_g729
codec,GSM,mod_voipcodecs
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_voipcodecs
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
25 total.
On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Mark,
I confirm that, as I wrote that wiki page (the early negotiation part) :)
Can you really confirm your FS version ?
The parameter you showed is old.
codec-prefs has been replaced in SIP profiles by:
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early
negotiation default):
http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile:
<param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki
page, I see the following sentence, which I interpret that GSM should
still be sent:
When FS calls leg B, the list of codecs in outbound-codec-prefs for
the SIP profile is reorganized by pushing the codec negotiated above
for leg A at the top . If B does not accept any of the codecs, the
calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano
<tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you
need to change that endpoint to also accept G711 or also license G729
in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of
codecs that are offered to an extension? I know I can change the
order in the codec_prefs, but would prefer FS to offer all three
codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
Thanks
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This looks like you have disable-transcoding set to true. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 30/06/2010 à 15:07, Mark Campbell-Smith a écrit :
Updating the configuration did not help.
I'm not telling FS to use any other codecs other than what I have specified below, so I'm not sure what I have done wrong. Below is the full trace:
------------------------------------------------------------------------ INVITE sip:1020@192.168.1.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae From: 1000 <sip:1000@192.168.1.120>;tag=2c7a518d12f9370eo0 To: <sip:1020@192.168.1.120> Call-ID: 316156b9-f6b413d2@192.168.1.121 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6- d9ad5b204ca2",uri="sip: 1020 @192.168.1.120 ",algorithm = MD5 ,response = "c4a5e5cb60676366b8bcfb1329f3fc08 ",qop=auth,nc=00000001,cnonce="6a0220d0" Contact: 1000 <sip:1000@192.168.1.121:5060> Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 451 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp
v=0 o=- 17789112 17789112 IN IP4 192.168.1.121 s=- c=IN IP4 192.168.1.121 t=0 0 m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
------------------------------------------------------------------------ send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684:
------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae From: 1000 <sip:1000@192.168.1.120>;tag=2c7a518d12f9370eo0 To: <sip:1020@192.168.1.120> Call-ID: 316156b9-f6b413d2@192.168.1.121 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 14-49-15 -0500 Content-Length: 0
------------------------------------------------------------------------ 2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121 Rejected by acl "domains". Falling back to Digest auth. 2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel sofia/internal/1000@192.168.1.120 [77a3ab82-8447-11df-93a7-d9ad5b204ca2] 2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_NEW 2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/1000@192.168.1.120) State NEW 2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel sofia/internal/1000@192.168.1.120 entering state [received][100] 2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP: v=0 o=- 17789112 17789112 IN IP4 192.168.1.121 s=- c=IN IP4 192.168.1.121 t=0 0 m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30
2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[G729:18:8000:20] 2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[PCMU:0:8000:20] 2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec Compare [PCMU:0:8000:30]/[GSM:3:8000:20] 2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting codec PCMU@30i@8000h 2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec sofia/internal/1000@192.168.1.120 PCMU/8000 30 ms 240 samples 2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf send/recv payload to 101 2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451 (sofia/internal/1000@192.168.1.120) State Change CS_NEW -> CS_INIT 2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000@192.168.1.120 [BREAK] 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_INIT 2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000@192.168.1.120) State INIT 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83 sofia/internal/1000@192.168.1.120 SOFIA INIT 2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117 (sofia/internal/1000@192.168.1.120) State Change CS_INIT -> CS_ROUTING 2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000@192.168.1.120 [BREAK] 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/1000@192.168.1.120) State INIT going to sleep 2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_ROUTING 2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474 (sofia/internal/1000@192.168.1.120) Callstate Change DOWN -> RINGING 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000@192.168.1.120) State ROUTING 2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333 (sofia/internal/1000@192.168.1.120) Callstate Change RINGING -> ACTIVE 2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140 sofia/internal/1000@192.168.1.120 SOFIA ROUTING 2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1000@192.168.1.120 Standard ROUTING 2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing 1000->1020 in context default Dialplan: sofia/internal/1000@192.168.1.120 parsing [default->Local_1000_1019] continue=false Dialplan: sofia/internal/1000@192.168.1.120 Regex (FAIL) [Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1000@192.168.1.120 parsing [default->Mobile_1020s] continue=false Dialplan: sofia/internal/1000@192.168.1.120 Regex (PASS) [Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/ break=on-false Dialplan: sofia/internal/1000@192.168.1.120 Action set(dialed_extension=1020) Dialplan: sofia/internal/1000@192.168.1.120 Action export(codec_string=GSM) Dialplan: sofia/internal/1000@192.168.1.120 Action set(codec_string=GSM) Dialplan: sofia/internal/1000@192.168.1.120 Action bridge(user/${dialed_extension}@${domain}) Dialplan: sofia/internal/1000@192.168.1.120 Action set_user(1000@$ {domain}) Dialplan: sofia/internal/1000@192.168.1.120 Action answer() Dialplan: sofia/internal/1000@192.168.1.120 Action sleep(1000) Dialplan: sofia/internal/1000@192.168.1.120 Action system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount} ${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$ <action application=) 2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1000@192.168.1.120) State Change CS_ROUTING -> CS_EXECUTE 2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000@192.168.1.120 [BREAK] 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/1000@192.168.1.120) State ROUTING going to sleep 2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000@192.168.1.120) Running State Change CS_EXECUTE 2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/1000@192.168.1.120) State EXECUTE 2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233 sofia/internal/1000@192.168.1.120 SOFIA EXECUTE 2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1000@192.168.1.120 Standard EXECUTE EXECUTE sofia/internal/1000@192.168.1.120 set(dialed_extension=1020) 2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843 sofia/internal/1000@192.168.1.120 SET [dialed_extension]=[1020] EXECUTE sofia/internal/1000@192.168.1.120 bridge(user/1020@mydns.dyndns.org ) 2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956 variable string 0 = [presence_id=1020@mydns.dyndns.org] 2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel sofia/internal/sip:1020@192.168.1.123:5060 [77d7e3b6-8447-11df-93a8-d9ad5b204ca2] 2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883 (sofia/internal/sip:1020@192.168.1.123:5060) State Change CS_NEW -> CS_INIT 2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/sip:1020@192.168.1.123:5060 [BREAK] 2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/sip:1020@192.168.1.123:5060) Running State Change CS_INIT 2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/sip:1020@192.168.1.123:5060) State INIT 2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83 sofia/internal/sip:1020@192.168.1.123:5060 SOFIA INIT send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944:
------------------------------------------------------------------------ INVITE sip:1020@192.168.1.123:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ Max-Forwards: 69 From: "1000" <sip:1000@192.168.1.120>;tag=Z51Qve55SUHta To: <sip:1020@192.168.1.123:5060> Call-ID: 4f442091-feea-122d-448a-00e04c0312e9 CSeq: 132833080 INVITE Contact: <sip:mod_sofia@192.168.1.120:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19 14-49-15 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 246 X-FS-Support: update_display Remote-Party-ID: "1000" <sip:1000@192.168.1.120>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120 s=FreeSWITCH c=IN IP4 192.168.1.120 t=0 0 m=audio 27386 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:30
It fails after this with Not Acceptable Here / INCOMPATIBLE_DESTINATION
On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone <david.ponzone@gmail.com
wrote: Please, retry with a genuine config (the default one would be a wise choice). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :
Hi All,
I'm not really sure if I got a firm answer for this one. Is the only way to ensure that transcoding is performed is by using late-negotiation? Why aren't all my codecs sent in the INVITE message to the B-leg (extension 1020)?
Thanks!
On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
FS version and codecs are shown below, but my config file are probably
quite old. But I guess they should still work?
All codecs are loaded, and the call works if late negotiation is set
on profile internal.
As I wrote above:
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
I forgot to write that Extension 1000 does not support GSM (I want to
force transcoding). Is that why FS is filtering out GSM on the b- leg?
freeswitch@internal> version
FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
freeswitch@internal> show codecs
type,name,ikey
codec,ADPCM (IMA),mod_voipcodecs
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_voipcodecs
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_voipcodecs
codec,G.726 16k (AAL2),mod_voipcodecs
codec,G.726 24k,mod_voipcodecs
codec,G.726 24k (AAL2),mod_voipcodecs
codec,G.726 32k,mod_voipcodecs
codec,G.726 32k (AAL2),mod_voipcodecs
codec,G.726 40k,mod_voipcodecs
codec,G.726 40k (AAL2),mod_voipcodecs
codec,G.729,mod_com_g729
codec,GSM,mod_voipcodecs
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_voipcodecs
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
25 total.
On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone@gmail.com
wrote:
Mark,
I confirm that, as I wrote that wiki page (the early negotiation part) :)
Can you really confirm your FS version ?
The parameter you showed is old.
codec-prefs has been replaced in SIP profiles by:
<param name="inbound-codec-prefs" value="$$ {global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$$ {outbound_codec_prefs}"/>
David Ponzone Direction Technique
email: david.ponzone@ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
Check this good wiki page for how FS negotiates codecs (early
negotiation default):
http://wiki.freeswitch.org/wiki/Codec_Negotiation
I have this set in my internal profile:
<param name="codec-prefs" value="$${global_codec_prefs}"/>
and as stated before vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
Setting late negotiation works (thanks Sergey), but reading the wiki
page, I see the following sentence, which I interpret that GSM should
still be sent:
When FS calls leg B, the list of codecs in outbound-codec-prefs for
the SIP profile is reorganized by pushing the codec negotiated above
for leg A at the top . If B does not accept any of the codecs, the
calls fails, obviously.
On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano
<tgraziano@myitdepartment.net> wrote:
I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
Wouldn't fs only get involved when its media server is referred to?
If the "other endpoint" will only accept G729, doesn't that mean you
need to change that endpoint to also accept G711 or also license G729
in FS?
On 6/23/10, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Test Setup:
vars.xml:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
The call setup is extension 1000 calls extension 1020
1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
FS as the A-leg codec
2. Extension 1020 only supports GSM codec. The call fails with Not
Acceptable Here.
FS only offers G729 and PCMU to 1020. How do I change the number of
codecs that are offered to an extension? I know I can change the
order in the codec_prefs, but would prefer FS to offer all three
codecs to an extension.
m=audio 23662 RTP/AVP 0 18 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
Thanks
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Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding -- Sent from my Android phone with K-9 Mail. Please excuse my brevity.
can you send us the output of: sofia status profile internal (of course, hide IPs and else if required). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 30/06/2010 à 15:46, Mark Campbell-Smith a écrit :
Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding
-- Sent from my Android phone with K-9 Mail. Please excuse my brevity.
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Sure David. thanks for helping me with this. I have not included the registrations, as I didn't think that was necessary .... freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xx.xx.xx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xx.xx.xx URL sip:mod_sofia@192.168.1.120:5060 BIND-URL sip:mod_sofia@192.168.1.120:5060 TLS-URL sip:mod_sofia@192.168.1.120:442 TLS-BIND-URL sips:mod_sofia@192.168.1.120:442;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G729,PCMU,GSM CODECS OUT G729,PCMU,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 40 FAILED-CALLS-IN 8 CALLS-OUT 18 FAILED-CALLS-OUT 11 Registrations: 2010/6/30 David Ponzone <david.ponzone@gmail.com>:
can you send us the output of: sofia status profile internal (of course, hide IPs and else if required). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 30/06/2010 à 15:46, Mark Campbell-Smith a écrit :
Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding
-- Sent from my Android phone with K-9 Mail. Please excuse my brevity.
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in your dialplan add {absolute_codec_string=GSM} to your url <action application="bridge" data="{absolute_codec_string=GSM}sofia/internal/test@foo.com"/> On Wed, Jun 30, 2010 at 5:15 PM, Mark Campbell-Smith < mcampbellsmith@gmail.com> wrote:
Sure David. thanks for helping me with this. I have not included the registrations, as I didn't think that was necessary ....
freeswitch@internal> sofia status profile internal
================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xx.xx.xx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xx.xx.xx URL sip:mod_sofia@192.168.1.120:5060 BIND-URL sip:mod_sofia@192.168.1.120:5060 TLS-URL sip:mod_sofia@192.168.1.120:442 TLS-BIND-URL sips:mod_sofia@192.168.1.120:442;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G729,PCMU,GSM CODECS OUT G729,PCMU,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 40 FAILED-CALLS-IN 8 CALLS-OUT 18 FAILED-CALLS-OUT 11
Registrations:
2010/6/30 David Ponzone <david.ponzone@gmail.com>:
can you send us the output of: sofia status profile internal (of course, hide IPs and else if required). David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 30/06/2010 à 15:46, Mark Campbell-Smith a écrit :
Nope. I have transcoding enabled. At least I have that line disabing transcoding commented out in internal.xml and I assume by default FS enables transcoding
-- Sent from my Android phone with K-9 Mail. Please excuse my brevity.
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participants (6)
-
Anthony Minessale
-
Brian West
-
David Ponzone
-
Mark Campbell-Smith
-
Sergey Okhapkin
-
Tony Graziano