Archived from groups: comp.dcom.voice-over-ip (More info?)
I have seen issues in my testing with Cisco & Pingtel SIP phones,
where RTP codec preferences are defined & negotiated properly, but the
payload size for the codec is not. In the SIP messages, the size
(psize field I think) is not filled in. The Cisco phones want
increments of 10ms used for G.711, but my end may send 5, 15, 25, etc.
ms if it's not told that the other end will not handle these. This
causes voice failures in the call.
Has anybody else run across this issue, and any idea how or when it
will be resolved?
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