El mensaje (que comenzó el 5 de Agosto del 2005) dice:
I added TCP support to asterisk SIP channel. I put all my changes under #ifdef SIP_TCP_SUPPORT and left the original code. So if you search SIP_TCP_SUPPORT, you can find my changes very easily.
My changes
-Added TCP listening socket, siptcpsock.
-Added securechannel, sockfd, transport field to struct sip_pvt.
-Added transport, tcpsockfd field to struct sip_peer.
-Added TCP read in sipsock_read().
-Added siptcp_accept() to accept an incoming TCP connection request.
-Added transport, q parameter processing in Contact header parsing.
-Changed the hard-coded «UDP» in Via header to copy sip_pvt.transport.
-Added tcp_conenct() to make a TCP connection for outgoing message.
-Added TCP transmit in __sip_xmit().
-Saved TCP connecton socket to sip_peer.tcpsockfd, copied it to sip_pvt.sockfd when OPTIONS or INVITE is sent to the peer that is using TCP.
I tested it mainly xlite(UDP only free version) and Jain-SIP communicator. call signal is working well. One problem I am having is Jain-SIP communicator doesn’t receive any audio, I don’t know why. If any one has xlite-pro(TCP supported commercial version) or TCP supported SIP clients, I am looking forward to hear the test result.
Welcome any comment.
Thanks
y poco a poco han ido aportando su granito de arena para conseguir enviar el protocolo SIP a través de TCP con las ventajas que ello conlleva.
Enlace: http://bugs.digium.com/view.php?id=4903
Supongo que esperar a que salga en la versión de Asterisk 1.4.1 es quizá dificil, pero pronto, muy pronto…