GStreamer RTP Audio Pipe line issue

[Venkatesh Adiga]

Dear Group,I have been looking for the suitable gstreamer debug for one way audio issue. But  I could not succeed.  I could see that GST DEBUG log that "minimum latency bigger than maximum latencý" error was thrown out in case of error scenario.If anyone knows more about the error of latency, please throw some light.  Thank you RegardsVenkatesh 

    On Sunday, 24 January 2016 5:51 PM, Venkatesh Adiga <rv_adiga at yahoo.com> wrote:
 

 Dear Group,


 

    On Friday, 15 January 2016 6:39 AM, Venkatesh Adiga <rv_adiga at yahoo.com> wrote:
 

 Hi,
I am working on a Gstreamer based(for Audio Streaming in RTP Session) Media Gateway Development project. This is my first project that I got involved in Gstreamer based project, and have less knowledge on this domain.Below is the scenario of a Q.SIG to SIP call.Using the deskphone, the Caller A calls callee B's mobile. As per call signalling is concerned, the call is established between both the parties and RTP ports are activated at Media Gateway. RTP ports are opened and verified by using netstat command on a Linux box. The issue  that I am observing is that oneway audio was seen from callee to caller.As per wireshark trace, I am observing that Media Gateway performs the RTP packet transfer to individual call owner, but not forwarding from Caller or Callee or visa versa.
User A  <---> MG <----> User B - Treating User B and User A as seperate user in a call.
when user A speaks, Media Gateway should have done like the below.  User A ----->MG                            ------> User B
Could anyone throw some light on how to analyze the RTP issue in the above scenario of Gstreamer?What debug tools are available to verify Gstreamer pipeline in live call scenario?Thank you 
RegardsVenkatesh

   

  
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