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Hello, I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved. Thanks, j

...lplan will be very simple - extensions calling extensions within a context. No DTMF, no recording, no mixing for conference, etc. Cheers, j -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150324/fbe3a969/attachment.html>