Posted: Fri Feb 25, 2005 9:02 am Post subject: [Asterisk-Users] Speex transcoding for Cisco / Polycom
Hi guys,
I have a weird problem, and I have encountered a few other people with
the same issue. The problem is this:
Whenever I make a call from my IAXy (g711ulaw) to my server, and then my
server transcodes to speex and sends it to a remote asterisk server,
audio is perfectly fine. The same goes if I use Linphone with speex.
However, whenever I use a Cisco 7960 SIP 6.2 or a Polycom IP 500 with
SIP 1.3.1.0056, both using g711ulaw to my server, when it is transcoded
to speex and a connection is made to the remote system, audio chops and
I lose about 80% of all audio. This has been tested on asterisk 1.0.3,
1.0.4, 1.0.5, and CVS v1-0 from yesterday, all operate identically. The
problem works identically when I go to the remote and call to my server
with it transcoding from the Polycoms to my server.
If anybody has any insight to this problem, it would be appreciated.
(BTW, Speex 1.1.6 is just that, unstable... it will even crash asterisk
when you try to do "show translation recalc").
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