Parallel call forking case about webrtc-to-sip

[havfo]

I would probably add a branch flag to the different branches. I have made a gist with some changes to the existing config here.

The changes i have made are in:
route[WEB_SIP] and route[NATMANAGE].

In the rtpengine_manage calls in NATMANAGE, you could add the via-branch parameter to apply it to the correct one. I have not used the via-branch flag myself, but i guess via-branch=auto should work.

Note: the config does not run as is, it is only an example to explain.

[havfo]

And of course you would have to restructure NATMANAGE to your needs.

[korayvt]

The tricky part is what via-branch value should be used.

After some testing I realized that 'auto' option is not working. There is no code implementing this parameter.

Moreover all forked branches have the same via branch id which is the incoming INVITE's via branch id on branch route. So the values 1 and 2 to via-branch option is not functional in this case.

Therefore we have only 'extra' option with extra_pv_id variable.
And the question is what value do we need to set to this variable in requests on branch route and corresponding responses on reply route?

[havfo]

I have never had the need to set different branches myself, so not sure how much help I can be here.

In the example config above, the flag for SDES/DTLS is set pr. branch. So I guess you do something like:

if (isbflagset(FLB_SDES)) {
   $avp(extra_id) = "sdes";
   rtpengine_manage("...via-branch=extra...");
}

[havfo]

You might find some inspiration from this forum post.

[havfo]

You inspired me to update the kamailio config. It should now be able to handle different branches with different bridging cases.