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RTP player playback timing options

asked 2022-10-28 07:59:25 +0000

benw gravatar image


We are running a small communal network primarily based on Ubiquiti wifi wireless equipment. Some of our clients have been complaining about poor audio quality on their VOIP calls.

From the wireshark traces I have noticed that some packets are being buffered, causing a gap in communication (up to 300 ms) followed by followed by all packets immediately after one another.

In the RTP player I saw three options for Playback timing - Jitter buffer, RTP timestamp and uninterrupted mode. Which one should be used to give the most representative behavior of a VOIP phone? (I am using an Asterisk server, installed at the client, which plays a continuous playback)

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answered 2022-10-28 09:02:28 +0000

Jaap gravatar image

From the looks of it "Jitter buffer", but then again it depends on the time setting on that buffer vs. what (possible dynamic) jitter buffer setting the VoIP endpoint has, and what other measures the VoIP endpoint has for packet loss concealment.

You should have a look at the Pre-Conference Class II: Troubleshooting Voice over IP with Wireshark at SharkFest 22'EU next week.

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Asked: 2022-10-28 07:59:25 +0000

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Last updated: Oct 28 '22