I am Maimun, I would like to know how to configure RTP over TCP? Is it possible on Asterisk?
I want to analyse performance RTP over TCP. and compare it with performance RTP over UDP.
I have try SIP Signalling over TCP and succeed.
How about if we want RTP over Tunneling (SIPTunnel)?. Have somebody try it?
I found interesting issues siptunnel.sourceforge.net/, but I dont know how configure VoIP client and also SIP Server (Asterisk) to use it. no documentation on link.
or is there application which can convert RTP over UDP to TCP?
Then write and test the code to support it. Make sure that you have the right to donate it (in most places, this will require permission from your employer) and contribute it as a feature request with a patch.
Also, there are very good technical reasons why RTP runs over UDP, which actually bear on why RTP was invented in the first place.
With TCP, if you drop a packet, everything has to stop until the recovery mechanism can come into play, but with UDP, a lost or late packet can be interpolated.
TCP would only help with fax if the connection was VoIP from modec to modec, but in that case it would be more efficient to simply send the raw G4 encoded file using a file transfer protocol, or HTTP. If there is an analogue circuit, the retransmissions delays would completely disrupt the timing. Fax data is transmitted synchronously (although it would be possible to buffer lines and add padding).