How to force UDP transport on outgoing INVITE on asterisk SIP client


Even that i am using TCP as a transport, when i want to send INVITE during the call, it is going via UDP transport.
Is there any way to force it like TCP?

You need to print a SIP trace and your configuration. There is too little info to help you.

SIP trace is quite usual, REGISTER via TCP 5060, but when i call there, and want to DIAL, then INVITE message is sent with UDP as a transport, that’s the problem. More i need the info if somewhere i can configure it to let INVITEs come via TCP.


In sip.conf or pjsip.conf in case you have a normal system. Otherwise you have to check the files that are included by sip.conf or pjsip.conf.