I realize this isn't strictly Sofia-SIP but it affects the RTP side of
things built on Sofia.
I've tried two windows clients, SJPhone and X-Lite and both lie about their
bitrate. Both are advertising a PCMU/8000 codec. I watched the RTP packets
in a sniffer and ran the numbers: SJPhone is actually sending 8400 bits per
second (52.5 packets per second) and X-Lite is sending 7800 bits per
second. Both are running on the same windows box with a cheap embedded dell
sound board. The RTP stream with 8400 bits per second is causing a nasty 5%
audio latency when discontinuous transmission is turned off; the latency
grows at a rate of 3 seconds per minute since the windows client is sending
20ms worth of audio every 19.05ms. The extra buffering is eventually fatal
to the audio stream on my N770 when the DSP runs out of buffers (I think).
I've also seen similar affects from the only other windows box I've ever
tried but didn't have tcpdump handy to run the numbers myself. Has anyone
ever seen anything like this? I've been told that getting short accurate
audio samples with the Windows APIs but can't confirm that.
I'm looking at dynamically calculating a RTP streams real bitrate and
resampling it if it's too far off the claimed bitrate. That seems rather
kludgy.
thanks,
.mike
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