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I have established a SIP session between two clients. I observed the RTP trace between them on wireshark. The 'length' column has a value of 172 for the RTP packets that flow from Client 1 to Client 2 and has value 252 for packets that flow from Client 2 to Client 1. Client 1 and Client 2 are of different companies. Client 1 is running on a 64 bit Ubuntu 12.10 and Client 2 is running on Ubuntu 10.04. I would like to know what determines the length of the RTP packet.

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Probably a combination of codec, options like silence suppression, and the device doing the encoding.. just because it passes for an ubuntu server doesn't mean it is the originator of the RTP stream. – NickW Jun 6 '13 at 9:41

I can't give you any particular details, because as I said in my note, it depends on a large amount of factors (including codec, the presence of silence suppression, and sampling rate), but if you want some place to start, have a look at RFC3551.

Here are a few pertinent lines from it:

A G729 RTP packet may consist of zero or more G.729 or G.729 Annex A frames, followed by zero or one G.729 Annex B frames. The presence of a comfort noise frame can be deduced from the length of the RTP payload. The default packetization interval is 20 ms (two frames), but in some situations it may be desirable to send 10 ms packets.

The RTP timestamp clock rate is always 90,000, independent of the sampling rate. MPEG-1 audio supports sampling rates of 32, 44.1, and 48 kHz (ISO/IEC 11172-3, section 1.1; "Scope"). MPEG-2 supports sampling rates of 16, 22.05 and 24 kHz. The number of samples per frame is fixed, but the frame size will vary with the sampling rate and bit rate.

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Having a look at this paper could provide some answers:

Effect of Packetization on VoIP Performance

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