After some more analysis of my "dropped fragment" problem, things look
like this:
Cisco 7940 phone -- RTP --> chan_oh323 --> Asterisk
(running, eg., VoiceMailMain)
That RTP connection was negotiated via H.323 on a third machine running
Cisco CallManager 3.2, but this part should not be relevant.
Connections work fine, with one exception:
Whenever there's a break in *'s voice stream (eg. between the "mailbox"
and "password" prompts), the 7940 detects horrible jitter and drops a few
packets (eg. the whole "password" prompt).
Using ethereal, I found that the RTP packets sent by asterisk seem to have
bogus timestamps:
After the gap, timestamps continue just as if there hasn't been a gap, so
timestamp / sequence number always is constant.
This should be fine for continuous RTP streams, so I tried disabling
silence suppression in oh323.conf. However, * still only sends out packets
while it is playing, and not between playback phases.
So AFAICT, there are two possible solutions:
1) make chan_oh323 stream continuously, no matter if the current
application does not play audio. IOW: transmit silence instead of no
packets. Is this possible?
2) use better timestamps in streamed packets, ie increase timestamps even
after a period of silence, and not only for each sent packet.
Not sure if that makes the phone happy, though...
Any chance to do one of those?
Thanks in advance,
Siggi
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