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.

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.