Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
seems that if you get that log you didn't use jitetr buffer at all. In my opinion the latest jitter 1.2-branch is not working, the last working seems 1.2.1 patched. Hope Zoa could lead us to fix it. Regards Rosario - Original Message - From: Adam Moffett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 17, 2006 11:10 AM Subject: Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5 jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2 for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer I installed the jitterbuffer-1.2 branch and I have a few questions. First and foremost I'm getting hundreds of lines like this in my log file: Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 The console shows something similar: Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:10 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 My log file is going to be very big today. What could be responsible for frames (every frame?) having invalid timing info? Second I'm not sure if it's actually doing anything. For testing, I tried setting the max size to 2000ms and implementation to fixed.if I'm reading the comments in the sample config correctly that should create a 2000ms fixed jitter buffer, which in turn should mean a 2 second delay in audio, but I wasn't hearing any delay at all. Is this not a valid way to test whether the jitter buffer is doing something? ThirdI'm interested in a way to create some jitter ;) I was thinking I might take an ethernet hub and try to saturate it with several simultaneous large file transfers or something like that. Another possibility might be an 802.11 wireless connection at a fairly long range. If anyone knows of a more convenient way for me to create a jittery connection I'd be very interested. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2 for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer I installed the jitterbuffer-1.2 branch and I have a few questions. First and foremost I'm getting hundreds of lines like this in my log file: Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 Mar 17 10:54:03 WARNING[22831] abstract_jb.c: Recieved frame with invalid timing info: has_timing_info=0, len=1668178290, ts=1718447988 The console shows something similar: Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:09 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 Mar 17 10:57:10 WARNING[22866]: abstract_jb.c:347 ast_jb_put: Recieved frame with invalid timing info: has_timing_info=0, len=-1, ts=166273064 My log file is going to be very big today. What could be responsible for frames (every frame?) having invalid timing info? Second I'm not sure if it's actually doing anything. For testing, I tried setting the max size to 2000ms and implementation to fixed.if I'm reading the comments in the sample config correctly that should create a 2000ms fixed jitter buffer, which in turn should mean a 2 second delay in audio, but I wasn't hearing any delay at all. Is this not a valid way to test whether the jitter buffer is doing something? ThirdI'm interested in a way to create some jitter ;) I was thinking I might take an ethernet hub and try to saturate it with several simultaneous large file transfers or something like that. Another possibility might be an 802.11 wireless connection at a fairly long range. If anyone knows of a more convenient way for me to create a jittery connection I'd be very interested. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
14 mar 2006 kl. 15.38 skrev Matt: The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). Ok... so if I pull the 'jitterbuffer' branch I would get 'svn trunk code'. But if I pull 'jitterbuffer-1.2' I get the same code as I would have if I downloaded 1.2.5 and then applied a jitterbuffer patch (which I know, does not exist for 1.2.5). No, but you would get 1.2.5 + jitterbuffer patch + any changes to the 1.2 branch after we released 1.2.5. /Olle --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
14 mar 2006 kl. 19.00 skrev Robert Webb: On Tue, 14 Mar 2006 14:32:02 +0100 Olle E Johansson [EMAIL PROTECTED] wrote: 14 mar 2006 kl. 13.35 skrev Matt: Right saw that. But I'm trying to get away from using CVS-HEAD :) We all are. Every developer have switched from CVS to Subversion :-) This is not the development branch, but the release branch code, which we use to create the 1.2.x releases. The jitterbuffer itself is *not* release branch code, it's very much development. Please test it. The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). /O Olle, Pardon this dumb question please, but where are these test located. I looked under http://svn.digium.com and do not see them. I am not fluent in where everything is located and would like to do some testing on some of the other items such as the sip jitterbuffer. It will only be minimal but I would like to help where I can. For viewing: http://svn.digium.com/view/asterisk/team/oej - then pick a branch For checking out svn checkout http://svn.digium.com/svn/asterisk/team/oej/name name Use jitterbufferfor svn trunk + jitterbuffer jitterbuffer-1.2for 1.2 + jitterbuffer test-this-branchfor the test branch with a lot of cool stuff including the jitterbuffer Thanks for testing! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] - Asterisk Developer * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
Right saw that. But I'm trying to get away from using CVS-HEAD :) Is the jitterbuffer patch PURELY 1.2.5 with the patch in place? On 3/14/06, Olle E Johansson [EMAIL PROTECTED] wrote: 13 mar 2006 kl. 21.59 skrev Matt: Hi, I really want to start using 1.2.5, but I also really need to have the jitter buffer. Can anyone offer a suggestion of how to go? I've looked at the SIP_JB patch and there seems to be no indication of a patch for the 1.2.5 release. Look again. There is a new branch called jitterbuffer-1.2 that follows svn HEAD in the 1.2 branch. This is documented in the bug tracker report for the jitterbuffer :-) Please test! Thanks! /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
14 mar 2006 kl. 13.35 skrev Matt: Right saw that. But I'm trying to get away from using CVS-HEAD :) We all are. Every developer have switched from CVS to Subversion :-) This is not the development branch, but the release branch code, which we use to create the 1.2.x releases. The jitterbuffer itself is *not* release branch code, it's very much development. Please test it. The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). Ok... so if I pull the 'jitterbuffer' branch I would get 'svn trunk code'. But if I pull 'jitterbuffer-1.2' I get the same code as I would have if I downloaded 1.2.5 and then applied a jitterbuffer patch (which I know, does not exist for 1.2.5). Is that correct? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
On Tue, 14 Mar 2006 14:32:02 +0100 Olle E Johansson [EMAIL PROTECTED] wrote: 14 mar 2006 kl. 13.35 skrev Matt: Right saw that. But I'm trying to get away from using CVS-HEAD :) We all are. Every developer have switched from CVS to Subversion :-) This is not the development branch, but the release branch code, which we use to create the 1.2.x releases. The jitterbuffer itself is *not* release branch code, it's very much development. Please test it. The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). /O Olle, Pardon this dumb question please, but where are these test located. I looked under http://svn.digium.com and do not see them. I am not fluent in where everything is located and would like to do some testing on some of the other items such as the sip jitterbuffer. It will only be minimal but I would like to help where I can. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/ On 3/14/06, Robert Webb [EMAIL PROTECTED] wrote: On Tue, 14 Mar 2006 14:32:02 +0100 Olle E Johansson [EMAIL PROTECTED] wrote: 14 mar 2006 kl. 13.35 skrev Matt: Right saw that. But I'm trying to get away from using CVS-HEAD :) We all are. Every developer have switched from CVS to Subversion :-) This is not the development branch, but the release branch code, which we use to create the 1.2.x releases. The jitterbuffer itself is *not* release branch code, it's very much development. Please test it. The jitterbuffer branch is based on svn trunk (the same as the old CVS HEAD) The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD (meaning latest 1.2 version code). /O Olle, Pardon this dumb question please, but where are these test located. I looked under http://svn.digium.com and do not see them. I am not fluent in where everything is located and would like to do some testing on some of the other items such as the sip jitterbuffer. It will only be minimal but I would like to help where I can. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
On Tue, 14 Mar 2006 13:44:57 -0500 Matt [EMAIL PROTECTED] wrote: http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/ Thank you I was looking directly under asterisk and not team. :-) Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Jitter Buffer for 1.2.5
Hi, I really want to start using 1.2.5, but I also really need to have the jitter buffer. Can anyone offer a suggestion of how to go? I've looked at the SIP_JB patch and there seems to be no indication of a patch for the 1.2.5 release. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Jitter Buffer for 1.2.5
13 mar 2006 kl. 21.59 skrev Matt: Hi, I really want to start using 1.2.5, but I also really need to have the jitter buffer. Can anyone offer a suggestion of how to go? I've looked at the SIP_JB patch and there seems to be no indication of a patch for the 1.2.5 release. Look again. There is a new branch called jitterbuffer-1.2 that follows svn HEAD in the 1.2 branch. This is documented in the bug tracker report for the jitterbuffer :-) Please test! Thanks! /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
