Hello,
Could someone please kindly point me in the right direction to a TCP SIP patch for chan_sip.so etc.
I really want to implement Asterisk 1.4, but i’m desperate to get TCP SIP working.
Please please help! 
You can’t do SIP with TCP. Normal TCP operation is detrimental to real-time SIP audio.
I’ll give you two examples:
TCP (Transmission control protocol) transmits packets, and ensures that they arrive by retransmitting a packet if it doesn’t make it. (This is bad.) What would you do with an audio stream packet that was retransmitted to you, but arrived late, and out of order? Play it anyway? Of course not… you needed it 900ms ago. It’s usless now, so you’d drop it. You’re better off not having the packet retransmitted in the first place.
When network collisions are detected, TCP will also delay transmission, waiting until a resource becomes available (transmission backoff) making real-time transmission impossible. You can’t let something like TCP add to latency issues.
SIP uses UDP transmission. UDP does not do collision detecton, or retransmission. While UDP does not guarantee that packet will arrive at the intended destination, it’s about as close to real time transmission as Ethernet will allow.
Hi.
Why are you going to use TCP SIP?
SIP exhort mans to use UDP Protocol, but you can use TCP, SCTP and so on.
SIP say “I have made two default ports such as 5060 for TCP and UDP and 5061 for TLS over TCP”.
** SIP can use any network transport protocol.
If you are going to use TCP you can use next URIs I say.
URIs:
sip:kingstar_713@naenara.co.kp
sip:molla_222:secretword@192.168.0.222;transport=tcp
sip:blond_222@ours.com;transport=tcp
sip:+859-212-555-1212:222@ours.com;user=phone
sips:7041@bei.net
sip:itbt;day=tuesday@ours.com
sip:dotdot.com;method=REGISTER?to=haoxiang%40dotdot.com
As you can see above you can use TCP protocol.
If you want to GET the correct information, Look [b]RFC 3261/b, RFC 768 for UDP and RFC 761 for TCP.
I hope the work is improved for you.