Hi Siggi,
Thanks for your thorough report and test results. You are right. The transmission of voice packets after silence periods is done with incorrect timestamps, causing slight voice drop-outs. I 'll see how this can be fixed and let you know.
Regards, Michael.
Siggi Langauf wrote:
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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