After upgrading Asterisk from 1.0.9 to 1.2, the audio in MeetMe conferences is distorted. It’s VERY garbled. Audio sounded fine in version 1.0.9. Any ideas what could be causing this?
I did a make clean before compiling the new version. I’ve tried adjusting the audiobuffers option to no avail.
I have not noticeed this myself and I use a lot of meetme on my servers. I have seen a lot of issues pop up with H323 and also with SIP and certain codecs. It would help if you gave a little more information like:
what protocols and codecs clients are using to connect
what timer is being used by your system
how many participants and what machine load is at the time of the audio garbling
We are also experiencing problems with local callers being garbled to outside callers. As an experiment, we had our inside callers place an outside call back to our office and to the conference, and now they sound find. I suspect a timing issue.
Our inside callers are using SIP (sipura’s), and our outside calls are to a T1 using Digium hardware.
i’m having the issue as well. reading up on the bug, it appears that the main problem is packet sizes over 30ms cause the problem. to fix it, change your RTP packet size to .010 - on the SPA-841’s it’s set to .030 by default. not sure about other phones. to change it, click admin login, then advanced, then look for it under the sip tab.
still getting a bit of an echo when in meetme, though… i hear my own voice come back to me when i talk. others do as well.