Thanks guys, I will try to reproduce it and report in jira.<div><br></div><div>Cheers<br><br><div class="gmail_quote">On Tue, Mar 5, 2013 at 11:52 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">We have all the updates in sofia tree plus many, minus one I just committed to the sofia tree last weekend (that I'll pull in this week). That commit was the first commit to the sofia tree in 18 months.<div>
<br><div><br><div><div><div class="h5"><div>On Mar 5, 2013, at 10:27 AM, Ken Rice <<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>> wrote:</div><br></div></div><blockquote type="cite">
<div><div><div class="h5">
<font face="Monaco, Courier New"><span style="font-size:11pt">Open a Ticket in Jira... Include the debug information and make sure you are testing against current git master head...<br>
<br>
Replacing the sofia-sip we have in tree with 1.12.11 will do you no good, a) many of the patches in the last release actually came from FreeSWITCH, 2) we have many more patches since then to go into sofia-sip which you will be missing as 1.12.11 was released nearly 2 years ago.<br>
<br>
K<br>
<br>
On 3/5/13 4:18 AM, "Xijing Dai" <<a>dxj19831029@gmail.com</a>> wrote:<br>
<br>
</span></font></div></div><blockquote type="cite"><div><div class="h5"><font face="Monaco, Courier New"><span style="font-size:11pt">I had a very random sip error in the freeswitch, and I want to try sofia-sip 1.12.11 to see if it's fixed there, but there are a few build errors。<br>
<br>
Is there any race condition chance inside mod_sofia?<br>
<br>
We make a few calls from the same softphone simultaneously, it includes pause/resume/terminate on any call at any time.<br>
<br>
The outbound endpoint is going though loopback inside freeswitch.<br>
<br>
Sometimes, the paused inbound call won't be able to resume, and after some deep tracing inside freeswitch, it seems to me that the sip message for resuming call is handled on outbound non-loopback channel.<br>
And I used wireshark to catch the packets, it shows that the call is in trying stage.<br>
<br>
This happens very randomly.<br>
<br>
Do you guys have any idea?<br>
<br>
Cheers<br>
<br><br>
On Tue, Mar 5, 2013 at 4:57 PM, Steven Ayre <<a>steveayre@gmail.com</a>> wrote:<br>
</span></font></div></div><blockquote type="cite"><div><div class="h5"><font face="Monaco, Courier New"><span style="font-size:11pt">Is there a particular reason you're asking about Sofia-SIP 1.12.11? Are there any specific new bugfixes/features in that version you're after?<br>
<br>
-Steve<br>
<br>
<br>
<br>
<br>
On 5 March 2013 08:56, Steven Ayre <<a>steveayre@gmail.com</a>> wrote:<br>
</span></font></div></div><blockquote type="cite"><div><div class="h5"><font face="Monaco, Courier New"><span style="font-size:11pt">Sofia-SIP only provides the SIP signalling layer. FreeSWITCH uses its own RTP stack for media, and in the progress of moving to crtp.<br>
<br>
The current combination of libraries is well tested and stable, and as Peter points out contain many patches that are improvements over upstream.<br>
<br>
Because of this stability the developers aren't making any major updates to the library versions at the moment. Master (1.3) is the development branch for v1.2.stable. They're currently working towards the next stable 1.2 release, after which they plan to create a new 1.4 branch. 1.2 will continue to be 'stable' while in 1.4 they will start doing major updates to the libraries. That's likely to introduce a period of instability in 1.4 that you won't have if you remain on 1.2, the price for running on the bleeding edge.<br>
<br>
Is there a particular rea<br>
<br>
-Steve<br>
<br>
<br>
<br>
On 5 March 2013 08:06, Subhash <<a>gvvsubhashkumar@gmail.com</a>> wrote:<br>
</span></font></div></div><blockquote type="cite"><div><div class="h5"><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
Does sofia-sip supports rtp rfc also?<br>
<br>
On Mar 5, 2013 1:22 PM, "Peter Olsson" <<a>POlsson@enghouse.com</a>> wrote:<br>
</span></font></div></div><blockquote type="cite"><font face="Monaco, Courier New"><div><div class="h5"><font><span style="font-size:10pt">I believe that the Sofia library within the FS source tree has more updates than this version – most of the updates in the last official Sofia release came from the FS developers anyway. So going to 1.12.11 would probably be a downgrade.<br>
<br>
I’m not 100% sure about this though...<br>
<br>
/Peter<br>
<br>
<b>Från:</b> <a>freeswitch-dev-bounces@lists.freeswitch.org</a> [<a href="mailto:freeswitch-dev-bounces@lists.freeswitch.org" target="_blank">mailto:freeswitch-dev-bounces@lists.freeswitch.org</a>] <b>För </b>Xijing Dai<br>
<b>Skickat:</b> den 5 mars 2013 08:37<br>
<b>Till:</b> <a>freeswitch-dev@lists.freeswitch.org</a><br>
<b>Ämne:</b> [Freeswitch-dev] any plan to upgrade to Sofia-SIP 1.12.11 ?<br></span></font></div></div><span style="font-size:11pt"><br>
<br><div class="im">
Is there any plan to upgrade freeswitch to use Sofia-SIP 1.12.11 <<a href="http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt" target="_blank">http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.11.txt</a>> ?<br>
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