This is a big problem when investigating issues in load tests where the SIP traffic is too complex to follow.
I think we could use Call-Id that exists both in REGISTER and INVITE methods.
This is mainly for jain-sip-js, but maybe we could do that in webrtcomm connectors as well.
This is a big problem when investigating issues in load tests where the SIP traffic is too complex to follow.
I think we could use Call-Id that exists both in REGISTER and INVITE methods.
This is mainly for jain-sip-js, but maybe we could do that in webrtcomm connectors as well.