Tag: g729

  • RTP pass-through mode in Asterisk

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    What is RTP pass-through mode in Asterisk? The speech for VoIP calls uses RTP (Real Time Protocol) to get from one end to the other and it is compressed using one of the many speech compression codecs available. Commonly used codecs include G.711 and G.729. When you call someone through an Asterisk PBX, there will be two “legs” to that call. The first leg is from your phone to Asterisk and the second is from Asterisk to…

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Please note this article is old and
only relevant if using chan_sip.

If you are using the newer pjsip,
this article is not relevant.

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