Joseph Love wrote:
The issue which occurs is that the audio from the SIP client to the IAX
client will spend most of it's time sounded very robotic, and garbled.
It is possible, although very difficult to understand someone who is on
the SIP phone.
I have asterisk 1.2.10 configured with realtime with both IAX and SIP
clients.
The SIP clients include a Grandstream gxp2000 hard phone, and
Counterpath's X-Lite 3 (for windows) softphone.
The IAX clients tested include idefisk (both windows & mac), JakenIAX,
and LoudHush.
GSM is the preferred codec of both IAX & SIP clients, and is indeed the
codec being used in all tests.
Audio from the IAX to the SIP client does not experience any issues.
SIP to SIP (and presumably, although untested, IAX to IAX) communication
does not experience any issues.
We also have a T1 card through which many calls have been placed, both
from the IAX and SIP phones, without any audio issues occurring, in
either case.
If it weren't for that there have been multiple clients tested to verify
this robotic sound, I would cough it up to it being a incompatability
between the particular clients, but this occurs on all SIP-IAX
communication that has been tried.
I'm running out of options as SIP-IAX intercommunication is kinda
expected (and necessary for me), and out of good softphones for the mac,
as most of the mac-compatible softphones are IAX2-based.
Please let me know what additional information is needed to help me
debug this problem.
Can you try different codecs just to rule out any issues with that?
E.g., if both devices use ulaw, do you still have the same problem?
I've used both iaxcomm and x-lite to communicate with cisco, polycom,
grandsteam, etc, without that type of problem.
Is it possible to obtain an ethereal trace of both the iax and sip/rtp
streams in the same trace? If so, a couple hundred packets should be
more then enough to see what's going on.
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