[Asterisk-Users] filling suppressed silence with chan_oh323

[Michael Manousos]

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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