Fix sequence number used by asterisk for outgoing RTP packets (!136) · Merge requests · Voice / asterisk

Grzegorz Sluja requested to merge fix_sequence_number_srtp_7.2 into release-7.2 Nov 02, 2023

There was no audio for 3-way conference when sRTP is used. For 2-way calls frame->seqno is taken from DSP and is used by asterisk for the sequence number in RTP headers. However for 3-way conference the sequence number is generated by asterisk and it has to be greater than the previous value, otherwise libsrtp refuses to forward 'too old' RTP packets.

Fix sequence number used by asterisk for outgoing RTP packets

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