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SIP Audio Problems: Difference between revisions
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* Get a pcap and use Wireshark to decode it
* Wireshark can play audio with different jitter buffer settings so you can sometimes recreate audio problems from the pcap
* Wireshark can [
** Packets can arrive bunched up, with gaps between, or out of order. Jitter is a measurement of this.
** Wireshark calculates jitter as per the definition in [
** Delta is the time difference between arrival of RTP packets. Max delta is the highest gap between arrival of packets. If the max delta is greater than the jitter buffer, audio will start getting dropped. Each packet usually contains 20ms of audio, so for high enough max delta you'll start to lose audio in 20ms chunks.
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