[Asterisk-Users] SIP - NATIVE BRIDGE ERROR
Wes Marderness
wmarderness at sigmabit.com
Fri Feb 6 15:32:53 MST 2004
Hi,
Running Version 0.7.2, I receive the following error when attempting to
connect two SIP Devices.
WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 =
524558, cannot native bridge.
The bridge is made but the quality of the call is bad, a lot of disturbing
noises in background.
Oddly enough, both devices are using only one codec G729. I also am using
the demo G729 license for Asterisk. I'm not sure how 2 different codecs are
being found.
I saw in ast_rtp_bridge function, that the get_codec function returned these
values. Could anyone tell me where the get_codec function is? Curious as to
how this is happening.
Should this problem be added to the bug tracker? The SIP calls are very bad,
and I did not experience this problem with 0.5.0 .
Thanks,
Wes
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