Sofia late-negotiation on re-INVITE (codec-modification)
Basic call SIP vs SIP with: <param name="inbound-late-negotiation" value="true"/> <action application="set" data="inherit_codec=true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> This help to avoid FS transcoding, the SIP endpoints negotiate codec end-to-end. FS still in the media path but no transcoding applied. INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] If a SIP ep send a re-INVITE for codec modification, sofia negotiate the new codec on that leg without "asking" the remote party. This result in transcoding and since I have mod_g729 only in passthrough mode the call is released by the core. re-INVITE [g729] ---> FS 200 OK [g729] <--- FS ************* It seems that "late_negotiation + inherit_codec" doesn't apply to a codec modification. Do I miss some configuration options in the SIP profile? (Note: I cannot use proxy-media/no-media mode in my configuration) ************* Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------
You do realize that those param's are sofia profile params and not dialplan XML right? /b On Nov 25, 2011, at 12:27 AM, Visciano Giovanni wrote:
<param name="inbound-late-negotiation" value="true"/> <action application="set" data="inherit_codec=true"/> <param name="renegotiate-codec-on-reinvite" value="true"/>
You do realize that those param's are sofia profile params and not dialplan XML right?
Hi Brian, sure I know that. So: In sofia SIP profile I have: <param name="inbound-late-negotiation" value="true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> and in my dialplan XML I hit: <action application="set" data="inherit_codec=true"/> I repeated my test scenario, sorry what I described was with inherit_codec=false. If I set inherit_codec=true (this is my purpose) sofia behavior is this: INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] re-INVITE [g729] ---> FS Reinvite Codec Error! <--- FS Here the call is release and from the log I see that sofia just negotiatiate localy and compare the codec list in the new offer only with PCMA. With inherit_codec=false it compare the codec list with the codec loaded in the core so it match G729 (but damn it's a passthrought codec and I cannot do transcoding!). Here is an extract off the log, that show the error. Now I do not have with me the complete session log, I will send it tomorrow. Do you think it's a bug? 1970-01-01 23:10:28.600056 [DEBUG] switch_core_session.c:857 Send signal sofia/external/abcd@xyz.com [BREAK] 1970-01-01 23:10:28.600056 [DEBUG] switch_core_session.c:857 Send signal sofia/external/abcd@xyz.com [BREAK] 1970-01-01 23:10:28.620894 [DEBUG] sofia.c:5143 Channel sofia/external/abcd@xyz.com entering state [received][100] 1970-01-01 23:10:28.620894 [DEBUG] sofia.c:5154 Remote SDP: v=0 o=- 0 1 IN IP4 10.50.185.16 s=IMSS c=IN IP4 10.50.185.16 t=0 0 m=audio 10992 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 1970-01-01 23:10:28.620894 [DEBUG] sofia_glue.c:4731 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 1970-01-01 23:10:28.620894 [DEBUG] sofia_glue.c:4731 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 1970-01-01 23:10:28.620894 [DEBUG] sofia_glue.c:4845 Set 2833 dtmf send/recv payload to 101 1970-01-01 23:10:28.620894 [ERR] sofia.c:5611 Reinvite Codec Error! Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------
Finally we are back to our test. I updated my FS installation to last GIT (FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600)) TEST ---- SIP vs SIP basic audio call, then re-INVITE for codec modification. FS configuration B2B, avoid transcoding. CONF ---- In sofia SIP profile I have: <param name="inbound-late-negotiation" value="true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> and in my dialplan XML I hit: <action application="set" data="inherit_codec=true"/> Loaded codec modules: freeswitch@internal> show codec type,name,ikey codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.729,mod_g729 codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE TEST 1) ------- 1001 ----invite(pcma)----> FS --invite(pcma)--> 1000 1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000 1001 --re/invite(g729)---> FS 1001 <----488------------ FS Full log: http://pastebin.freeswitch.org/17906 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. From the log I see 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error! TEST 2) ------- 1000 ----invite(pcma)----> FS --invite(pcma)--> 1001 1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001 1000 FS <--re/invite(g729)- 1001 1000 FS ------------------> 1001 Full log: http://pastebin.freeswitch.org/17907 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding! I hate transcoding!
From the log I see 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA@20ms@8000hz to G729@20ms@8000hz 2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/1001@138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 sofia/internal/1001@138.132.110.64:5070 Push codec G729:18 2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error!
Regard, Giovanni PS: next week I won't be at work. My collegue Nevio will follow the problem, so if you need more data/details just tell us. Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------
Hi All, I've some issue, B send a re-invite changing the codec and fs hangup the call with 488 , Without reinvite a-leg. Regards, Alessandro Da: Visciano Giovanni <Giovanni.Visciano@italtel.it> Risposta: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Data: Thu, 1 Dec 2011 16:35:54 +0100 A: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Oggetto: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) R: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification) Finally we are back to our test. I updated my FS installation to last GIT (FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600)) TEST ---- SIP vs SIP basic audio call, then re-INVITE for codec modification. FS configuration B2B, avoid transcoding. CONF ---- In sofia SIP profile I have: <param name="inbound-late-negotiation" value="true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> and in my dialplan XML I hit: <action application="set" data="inherit_codec=true"/> Loaded codec modules: freeswitch@internal> show codec type,name,ikey codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.729,mod_g729 codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE TEST 1) ------- 1001 ----invite(pcma)----> FS --invite(pcma)--> 1000 1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000 1001 --re/invite(g729)---> FS 1001 <----488------------ FS Full log: http://pastebin.freeswitch.org/17906 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. From the log I see 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error! TEST 2) ------- 1000 ----invite(pcma)----> FS --invite(pcma)--> 1001 1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001 1000 FS <--re/invite(g729)- 1001 1000 FS ------------------> 1001 Full log: http://pastebin.freeswitch.org/17907 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding! I hate transcoding!
From the log I see 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA@20ms@8000hz to G729@20ms@8000hz 2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/1001@138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 sofia/internal/1001@138.132.110.64:5070 Push codec G729:18 2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error!
Regard, Giovanni PS: next week I won't be at work. My collegue Nevio will follow the problem, so if you need more data/details just tell us. Internet Email Confidentiality Footer **************************************************************************** **************************************************************** La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. **************************************************************************** **************************************************************** _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
I've opened a BUG. http://jira.freeswitch.org/browse/FS-3739 I don't know what's you FS configuration. What I know is: - SIP "media proxy mode" is OK. - re-INVITE for codec modification to image T.38 (t38-passthrough) is OK in all SIP "media mode". The "late_negotiation" + "inherit_codec" trick to avoid transcoding in a Back2Back (that is no media proxy) configuration works only for the very first call setup SDP offer/answer. http://wiki.freeswitch.org/wiki/Codec_negotiation Once the call is established, further codec modification via re-INVITE are usually closed on each leg (except for t38) and this can led to transcoding (and obvious errors if you can't or don't want to transcode). Hope this helps you find out your problem. Giovanni -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org on behalf of Alessandro Illiano Sent: Wed 12/21/2011 1:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) Hi All, I've some issue, B send a re-invite changing the codec and fs hangup the call with 488 , Without reinvite a-legS. Regards, Alessandro Da: Visciano Giovanni <Giovanni.Visciano@italtel.it> Risposta: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Data: Thu, 1 Dec 2011 16:35:54 +0100 A: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Oggetto: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) R: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification) Finally we are back to our test. I updated my FS installation to last GIT (FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600)) TEST ---- SIP vs SIP basic audio call, then re-INVITE for codec modification. FS configuration B2B, avoid transcoding. CONF ---- In sofia SIP profile I have: <param name="inbound-late-negotiation" value="true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> and in my dialplan XML I hit: <action application="set" data="inherit_codec=true"/> Loaded codec modules: freeswitch@internal> show codec type,name,ikey codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.729,mod_g729 codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE TEST 1) ------- 1001 ----invite(pcma)----> FS --invite(pcma)--> 1000 1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000 1001 --re/invite(g729)---> FS 1001 <----488------------ FS Full log: http://pastebin.freeswitch.org/17906 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. From the log I see 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error! TEST 2) ------- 1000 ----invite(pcma)----> FS --invite(pcma)--> 1001 1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001 1000 FS <--re/invite(g729)- 1001 1000 FS ------------------> 1001 Full log: http://pastebin.freeswitch.org/17907 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding! I hate transcoding!
From the log I see 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA@20ms@8000hz to G729@20ms@8000hz 2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/1001@138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 sofia/internal/1001@138.132.110.64:5070 Push codec G729:18 2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error!
Regard, Giovanni PS: next week I won't be at work. My collegue Nevio will follow the problem, so if you need more data/details just tell us. Internet Email Confidentiality Footer **************************************************************************** **************************************************************** La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. **************************************************************************** **************************************************************** _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
I don't think this is a bug... I think its by design. /b On Dec 21, 2011, at 11:26 AM, Visciano Giovanni wrote:
I've opened a BUG. http://jira.freeswitch.org/browse/FS-3739
I don't know what's you FS configuration. What I know is: - SIP "media proxy mode" is OK. - re-INVITE for codec modification to image T.38 (t38-passthrough) is OK in all SIP "media mode".
The "late_negotiation" + "inherit_codec" trick to avoid transcoding in a Back2Back (that is no media proxy) configuration works only for the very first call setup SDP offer/answer. http://wiki.freeswitch.org/wiki/Codec_negotiation
Once the call is established, further codec modification via re-INVITE are usually closed on each leg (except for t38) and this can led to transcoding (and obvious errors if you can't or don't want to transcode).
Hope this helps you find out your problem.
Giovanni
-- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian@freeswitch.org http://www.freeswitch.org
I used to say to my boss: ... it's not a bug it's a feature! I'm kidding :) By the way, even in this "design choice", al least the first test scenario is a bug (described in the first half on the Jira BUG): http://pastebin.freeswitch.org/17906 Moreover this "by design" behavior create some problem not only for transcoding issue but also for fax upspeed codec modification (no T38). Years ago, when hardware resources where poor, I had a lot of discussions with my colleagues to avoid the allocation of transcoding resources on our softswitch and we always ended up to use a transcoding resource if and only if it's really needed, if the endpoints can speak the same language let them do all the stuff. I know easy to say but ... Back to FS, what I think is we miss an endpoint independent codec negotiation signaling. I mean something like we have in the call setup phase via "absolute_codec_string" / "ep_codec_string" / SWITCH_ORIGINATOR_CODEC_VARIABLE. As I'm playing/prototyping a new endpoint module I used these channel variable to successfully interworking with SIP endpoint gateway in codec passthrough... but now that the test-suite is going over the SIP basic call codec modification I found that "by design problem". In the (working) spare time I'm trying to write a patch to extend the core interworking opening the way I suggested few lines over. As I'm very new to FS, for now it's a very embryonal and poor implementation in my opinion, but enough for the endpoint module prototype evaluation. I hope to give that back to you if I can find more time to move the patch to a presentable level. Giovanni -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org on behalf of Brian West Sent: Wed 12/21/2011 6:30 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sofia late-negotiation onre-INVITE(codec-modification) I don't think this is a bug... I think its by design. /b On Dec 21, 2011, at 11:26 AM, Visciano Giovanni wrote:
I've opened a BUG. http://jira.freeswitch.org/browse/FS-3739
I don't know what's you FS configuration. What I know is: - SIP "media proxy mode" is OK. - re-INVITE for codec modification to image T.38 (t38-passthrough) is OK in all SIP "media mode".
The "late_negotiation" + "inherit_codec" trick to avoid transcoding in a Back2Back (that is no media proxy) configuration works only for the very first call setup SDP offer/answer. http://wiki.freeswitch.org/wiki/Codec_negotiation
Once the call is established, further codec modification via re-INVITE are usually closed on each leg (except for t38) and this can led to transcoding (and obvious errors if you can't or don't want to transcode).
Hope this helps you find out your problem.
Giovanni
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Visciano: they have to be sofia profile params, because otherwise, the codec is already negotiated by the time it gets to the dialplan. -Avi On Sat, Nov 26, 2011 at 12:39 AM, Brian West <brian@freeswitch.org> wrote:
You do realize that those param's are sofia profile params and not dialplan XML right?
/b
On Nov 25, 2011, at 12:27 AM, Visciano Giovanni wrote:
<param name="inbound-late-negotiation" value="true"/> <action application="set" data="inherit_codec=true"/> <param name="renegotiate-codec-on-reinvite" value="true"/>
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participants (4)
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Alessandro Illiano
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Avi Marcus
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Brian West
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Visciano Giovanni