VC: My phone is broken: I get no audio. TAC: Show us a network trace. VC: (presents my ethereal traces, with the non-counting RTP timestamps) TAC: (laughing) NEXT!!!
I have not read RFC1889 (RTP) in detail, but I am positive that the
timestamp field was put there for a reason. Sure, maybe Cisco is a little
overzealous in the way their code handles non-conformance, but to try and put the blame entirely on them is misdirection. My ATA-186 has problems with the same RTP stream. GIGO.
* needs to generate RTP streams with valid timestamp progression -- surely
we're not happy to say "the Cisco 79x0 is the only phone that cares about
timestamps, so there's the problem".
Hi Vic,
For your information Sipura also suffers from the Timestamp issue. 3 months ago when I opened the case with them, they explained in detail why they needed those Timestamps (it has to do with the jitter buffer calculation algorithms). They told me the problem had to be solved at the Asterisk side since there is no reason why the Timestamps should change. They have not seen this weird behaviour with any other SIP system besides Asterisk. In any case, thats why we came up with the rtp.c hack, and have been happy ever since.
-- Andres Network Admin http://www.telesip.net
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