Created: (XMR-105) I get one way audio when a 183 or 180 (ringing) message arrives

I get one way audio when a 183 or 180 (ringing) message
arrives
------------------------------------------------------------
---

                 Key: XMR-105
                 URL: http://tra
ck.sipfoundry.org/browse/XMR-105
             Project: sipXtapi: sipXmediaLib
          Issue Type: Bug
          Components: sipXmediaLib
    Affects Versions: 2.9
         Environment: windows xp, build with visual studio
2003
            Reporter: Soledad Alborno
         Attachments: oneWayaudio.patch

The problem:
We are using the SipXtapi lib to dial-out to a SIP-to-PSTN
gateway. During the setup of the call we're receiving a
183+SDP from the gateway. This enables us to listen to the
dial-tone. (the one you usually hear before somebody picks
up the phone). This dial tone is generated by the gateway
(or somewhere further up in the network). After the 183+SDP
we're correctly receiving and sending media. After a few
seconds we receive a 180(RINGING) and the media is still
being send and received. When the call is answered on the
PSTN side we're receiving a 200+SDP. All goes well so far,
but imediately after receiving the 200+SDP the SipX library
stops sending media to the gateway. 

Simplified schematic drawing:

    A                B
0.0s       INVITE
    ----------------->
0.1s     100(TRYING)
    <-----------------
0.1s      183 + SDP
    <-----------------

          MEDIA
    <----------------> //both ways (silence from
gw-side)

3.0s    180(RINGING)
    <-----------------

          MEDIA
    <----------------> //both ways (dial tone from
gw-side)

        200 + SDP
5.0s <-----------------
           ACK
5.1s ----------------->

5.1s       MEDIA
    <----------------- //one way (audio from PSTN)

A = SipX client
B = SIP-to-PSTN gateway

We noticed this behavior when using the PlaceCall sample
application. 

The analysis:
When the 200 + SDP message is received by the voice client,
SipXCallLib executes the method
SipConnection::processInviteResponseNormal, then the
MpRtpOutputAudioConnection::handleStopSendRtp method is
called (from MpCallFlowGraph) to stop the channel that is
reproducing the ring tone. And then the method
MpRtpOutputAudioConnection::handleStartSendRtp is called in
order to start the outgoing audio channel which must send
audio to the other endpoint.

The problem is that
MpRtpOutputAudioConnection::handleStopSendRtp is adding to
the queue a message to deselect codecs and
MpRtpOutputAudioConnection::handleStartSendRtp does not put
a message in the queue, instead it calls to the handler
function directly

The solution:
in file sipXmediaLibsrcmpMpRtpOutputAudioConnection.cpp
in the method handleStartSendRtp delete ln 233 and 234
(written below)
      SdpCodec* codecs[] = {pPrimaryCodec, pDtmfCodec};

mpEncode->handleSelectCodecs(sizeof(codecs)/sizeof(SdpCod
ec*), codecs);
And add a line to enqueue the message to the encoder
      mpEncode->selectCodecs(pPrimaryCodec, pDtmfCodec);

PS:

I don't know if these 2.9 is the real version, I've checked
out the code from: http://scm.sipfoundry.org/rep/sipX/branches/sipXtapi/
and in a log from  PlaceCall I've found: sipXtapi SDK
2.9.1.0 Rls (built 0000-00-00)

This is the first time I works with SipX so maybe something
is wrong on the description of the solution. Please tell me
if something is wrong.

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Created: (XMR-105) I get one way audio when a 183 or 180 (ringing) message arrives

Thread: Created: (XMR-105) I get one way audio when a 183 or 180 (ringing) message arrives