<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-family: Calibri, sans-serif; "><div><div><div>Hi All,</div><div>I've some issue,</div><div>B send a re-invite changing the codec and fs hangup the call with 488 ,</div><div>Without reinvite a-leg….</div><div><br></div></div></div><div>Regards,</div><div>Alessandro</div><div><br></div><div><br></div><span id="OLK_SRC_BODY_SECTION"><div style="font-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt"><span style="font-weight:bold">Da: </span> Visciano Giovanni <<a href="mailto:Giovanni.Visciano@italtel.it">Giovanni.Visciano@italtel.it</a>><br><span style="font-weight:bold">Risposta: </span> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br><span style="font-weight:bold">Data: </span> Thu, 1 Dec 2011 16:35:54 +0100<br><span style="font-weight:bold">A: </span> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br><span style="font-weight:bold">Oggetto: </span> [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification)<br></div><div><br></div><div><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"><meta name="Generator" content="MS Exchange Server version 6.5.7654.12"><title>R: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification)</title><div><!-- Converted from text/plain format --><p><font size="2">Finally we are back to our test.<br>
I updated my FS installation to last GIT<br>
(FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600))<br><br>
TEST<br>
----<br>
SIP vs SIP basic audio call, then re-INVITE for codec modification.<br>
FS configuration B2B, avoid transcoding.<br><br>
CONF<br>
----<br>
In sofia SIP profile I have:<br>
<param name="inbound-late-negotiation" value="true"/><br>
<param name="renegotiate-codec-on-reinvite" value="true"/><br><br>
and in my dialplan XML I hit:<br>
<action application="set" data="inherit_codec=true"/><br><br>
Loaded codec modules:<br>
freeswitch@internal> show codec<br>
type,name,ikey<br>
codec,G.711 alaw,CORE_PCM_MODULE<br>
codec,G.711 ulaw,CORE_PCM_MODULE<br>
codec,G.729,mod_g729<br>
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE<br>
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE<br>
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE<br><br><br>
TEST 1)<br>
-------<br>
1001 ----invite(pcma)----> FS --invite(pcma)--> 1000<br>
1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000<br>
1001 --re/invite(g729)---> FS<br>
1001 <----488------------ FS<br><br>
Full log: <a href="http://pastebin.freeswitch.org/17906">http://pastebin.freeswitch.org/17906</a><br><br>
Note: FS does not negotiate end to end the reINVITE O/A codec modification.<br>
It is closed locally on the 1001->FS leg. From the log I see<br>
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]<br>
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]<br>
2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error!<br><br><br><br>
TEST 2)<br>
-------<br>
1000 ----invite(pcma)----> FS --invite(pcma)--> 1001<br>
1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001<br>
1000 FS <--re/invite(g729)- 1001<br>
1000 FS ------------------> 1001<br><br>
Full log: <a href="http://pastebin.freeswitch.org/17907">http://pastebin.freeswitch.org/17907</a><br><br>
Note: FS does not negotiate end to end the reINVITE O/A codec modification.<br>
It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding!<br>
I hate transcoding!<br>
>From the log I see<br>
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]<br>
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]<br>
2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA@20ms@8000hz to G729@20ms@8000hz<br>
2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec <a href="mailto:sofia/internal/1001@138.132.110.64">sofia/internal/1001@138.132.110.64</a>:5070 G729/8000 20 ms 160 samples 8000 bits<br>
2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 <a href="mailto:sofia/internal/1001@138.132.110.64">sofia/internal/1001@138.132.110.64</a>:5070 Push codec G729:18<br>
2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!<br>
2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error!<br><br>
Regard,<br>
Giovanni<br><br>
PS: next week I won't be at work. My collegue Nevio will follow<br>
the problem, so if you need more data/details just tell us.<br></font></p><p style="margin-top: 0; margin-bottom: 0">
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