[Asterisk-Users] RTP timestamps

[Andres]

>
>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