Voice breaks while using PJSIP and G729

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Hello everyone,

While receiving RTP G729 audio data, we are experiencing voice breaks in
the initial 4-6 secs. While analyzing the audio data, the voice breaks are
actually a flat line (zero amplitude) in time domain.

Here is a capture of time analysis of sample RTP G729 data with voice breaks.
http://www.signalogic.com/images/G729_rtp_payload_packets_that_cause_decoder_zero_output.jpg

Is there any explanation (theory) on the number of spikes generated in
decoded output?
In all the bad voice packets the first byte (78 hex) is the common
denomination. Does it imply something?
Is there any carrier, switch or telecom infrastructure that will instead
this kind or packets?

Currently the endpoint uses PJSIP stack with VOIPswitch as registrar server.

- Regards,
Pranav Desai



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