Zuñiga, Guillermo <guillermo.zun...@cwpanama.com> writes:
> Hi Fellows,
> 
> I am having  the following issue with one device that is having one way audio.
> 
> The Calls is from a SIP UA to PSTN site, when the call is stablishing
> the only way I have audio in both ways is when in the following case:
> 
> UA                                    SBC/GW->PSTN
> INVITE----------------->
> TRYING<----------------
> 183 W/SDP<------------
> RTP(RBT)<---------------
> RTP----------------------->(LIKE SILENCE FROM UA)
> 200OK<------------------
> ACK------------------->
> RTP<------------------------>RTP IN BOTH WAYS

You should note that the INVITE contains SDP, because a UA can send
an INVITE without SDP.

> Could you help me with this behavior of the UA is correct?

The behavior of the UA is *correct* relative to the standards, because
the standards only describe what the UA must send over the Internet.

But the behavior of the UA is *not useful* for your application.

The behavior is also *not common* for a UA.  Usually, a UA will render
to the user any RTP that it receives once the UA has sent an INVITE.

My recommendation is to request that the vendor of the UA improve the
UA, or to use a different UA.

> Other thing I am try to looking  for is that UA is failing with one
> way audio too is with the following scenarios.

> According with him, this could be related cause the GW is sending a
> 180WITHOUT SDP, that indicate that GW should not send early media or
> RBT, cause UA should play local ring back tone correct?, but RTP are
> coming from the GW cause I think PSTN is sending RBT.

The SDP in the INVITE tells the UAS(s) how to transmit RTP to the UAC.
Thus, the UAC does not need to receive SDP from the UAS in order to be
able to decode RTP sent by the UAS.

Again, the UAC is *permitted* to discard RTP that it receives, including
RTP that it receives before receiving SDP from the UAS.  But that
behavior is *not useful*.

Dale
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