Greetings List! I'm currently rolling out a new deployment of Asterisk 1.8 to replace existing 1.2 servers...and have run into an issue which could use your assistance! For testing I have trunked (iax2) two of the servers - one running 1.8 and the other at 1.2. Calls placed from SIP --> SIP sound fantastic and crystal clear. However, when I place a echo test call (*43) from 1.8 to 1.2 the result is a pulsing echo and garbled audio. The same result is found when dialing into a meetme conference being held on 1.2 from the 1.8 server. I know that the jitter buffer has gone through quite a bit of work since 1.2 but my results at this point seem to indicate 1.8 is actually worse than 1.2. I've also trunked together two 1.8 boxes (physical locations are in different countries), to rule out if 1.2 was somehow causing the extra jitter. My results were identical - echo testing/meetme from 1.8 to 1.8 resulted in the same audio issues. The new servers are in the same locations as our production units, using the same networks/etc... so I expected identical results between them, not for 1.8 to be substantially worse. Any recommendations? Has anyone else experienced a similar issue and if so, perhaps you could share your experience. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110908/68469f72/attachment.htm>
Danny Nicholas
2011-Sep-08 18:56 UTC
[asterisk-users] Jitter only affecting meetme - and echo testing
1. What version of 1.8 (known issues with some branches) 2. Why not 10.0? From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of JT Sent: Thursday, September 08, 2011 1:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Jitter only affecting meetme - and echo testing Greetings List! I'm currently rolling out a new deployment of Asterisk 1.8 to replace existing 1.2 servers...and have run into an issue which could use your assistance! For testing I have trunked (iax2) two of the servers - one running 1.8 and the other at 1.2. Calls placed from SIP --> SIP sound fantastic and crystal clear. However, when I place a echo test call (*43) from 1.8 to 1.2 the result is a pulsing echo and garbled audio. The same result is found when dialing into a meetme conference being held on 1.2 from the 1.8 server. I know that the jitter buffer has gone through quite a bit of work since 1.2 but my results at this point seem to indicate 1.8 is actually worse than 1.2. I've also trunked together two 1.8 boxes (physical locations are in different countries), to rule out if 1.2 was somehow causing the extra jitter. My results were identical - echo testing/meetme from 1.8 to 1.8 resulted in the same audio issues. The new servers are in the same locations as our production units, using the same networks/etc... so I expected identical results between them, not for 1.8 to be substantially worse. Any recommendations? Has anyone else experienced a similar issue and if so, perhaps you could share your experience. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110908/9d0da263/attachment.htm>