Ryan Huff wrote: > Yes, the handshake “noise” samples, so to speak (RTP payload) in a fax over > SIP will always be UDP; the signaling (SIP) could be TCP or UDP. If you do > have a poor connection, you could lose signaling due to packet loss, and the > connection terminate (whereas TCP would attempt a retransmission).
I think it's a myth that TCP's built-in retransmission mechanism somehow makes TCP SIP more reliable than UDP SIP. The truth is, for all SIP messages of any importance, the SIP protocol already has its own retransmission mechanism. Otherwise SIP would have always been TCP from its very beginnings. A lost UDP packet will be noticed by the sender not receiving any ACK message, who will then re-send. -mn _______________________________________________ cisco-voip mailing list [email protected] https://puck.nether.net/mailman/listinfo/cisco-voip
