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VoIP with Codec 729a: I understand the 8kbps sampling rate, but how do you come to have a 20 byte payload? IP/UDP/RTP header I understand, but with the codec, how do they determine they have a 20 or 40 byte payload with a 729a codec?
look at rfc deocuments
Easy.
All Codec use 8K bit/s sampling rate(PCM), but for different Codec, it generate different voice data size in different duration according to the Codec’s algorithm, for example :
G.723.1(5.3K) – generate 20 byte in 30ms, so the rate is (1000ms/30ms)*20byte*8bit=5280bit/s=5.3k
G.723.1(6.4k) – generate 24 byte in 30ms, so rate is (1000ms/30ms)*24byte*8bit=6336bit/s=6.4k
G.729A(8K) – generate 10 byte in 10ms, so rate is (1000ms/10ms)*10byte*8bit=8000bit/s=8k
You should have more clear understanding now, it’s hard to find answers from rfc documents for this.
Some supplement.
why sometimes you will see the G.729A have 20byte or 40byte pure data in one RTP package? because in this case the Voice Packetizer output the RTP data after two/four 10ms, so in fact it have 2/4 frames/package, it will save bandwidth when compare to 1 frame/package because now 2/4 frames share the same IP/UDP/RTP header;
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