[Kamailio-Users] Advanced Call Scenario / One Way Audio

[Brandon Armstead]

**resending as I am not sure if this made it out the first time, I do not
believe it did, if this is a duplicate -- my apologies.**

Hello,

   As always, thank you for all / any help and input you may provide in
advance.

Call Scenario:

UA1 -> REGISTRAR-01 -> Kamailio-01 -> Asterisk (New Call-ID + Asterisk in
Media Path) -> Kamailio-01 -> REGISTRAR-02 -> UA2

UA1 is behind NAT
UA2 is behind NAT

The purpose of this is when using a shared "USRLOC" database to simulate
calls from "PSTN" to generate both legs of the call, i.e. incoming and
outgoing, and also allow for easier / cleaner "traversal"

This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on
P1 and UA2 exists on P2) this prevents P1 -> UA2, and forces P2 -> UA2

We determine that this is a call from P1 to P2 (internal call) and thus
create this "bridge / interconnection"

We are running into a problem it seems with one way audio, i.e. the CALLEE
can hear the CALLER, however the CALLER CAN NOT hear the CALLEE.

REGISTRAR-01 AND REGISTRAR-02 are both "proxying" RTP

As well as the initial Asterisk in "the middle" SDP.

Let me know if this makes sense and if you guys have any further thoughts on
what may possibily be going wrong.

Perhaps there are better ways to go about this, let me know if I am way off
course, thank you!

Sincerely,
Brandon Armstead
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