AVTransmit2 example - I hear nothing

[zdenko961]

Hi all,

I am converting TAPI IVR application to JTAPI/JMF.

There are working CTI ports (tested with Cisco wave driver).

Test applications adds observers to CTI port and when I make call from device (7960), test application is notified and it answers call. As soon as call is answered "CiscoRTPOutputStartedEvImpl" event is sent. From event data I read host and port (Host=10.10.5.23,port=20360).

When I open phone's web page (http://10.10.5.23/) and go to stream statistic I see:

Remote Address 10.10.5.231/28000

Local Address 10.10.5.23/20360

Using AVTransmit2 class I start RTP session (AVTransmit2 is modified to use local port which is specified in CiscoMediaTerminal.register() method).

In log I can see:

Created RTP session Source: 10.10.5.231:28000

Created RTP session Dest: 10.10.5.23:20360

Network sniffer shows UDP packets but I hear nothing.

Wav file is 44100kHz, 16 bit, stereo.

When terminal was registerd CiscoMediaCapability is CiscoMediaCapability.G711_64K_30_MILLISECONDS

I suppose there is a problem with data format.

As I see from discusion example works, but it does not work for me.

Any sugestion how to solve proble?

Zdenko

[stephan.steiner]

Are you using the AVTransmit2 from the JMF pages at Sun, or the modified version I posted in this forum (G.711 streaming example)?

[zdenko961]

I am using sample you posted, with modification of localAddr:

sourcePort = 28000 + 2 * i; //specified in JTAPI initialization

localAddr = new SessionAddress( localHost, sourcePort);

[zdenko961]

Problem solved.

Problem was in wav file format. I tried with different file and it works.

"Bad" wav file was created using Windows sound recorder.

Zdenko

[Developer22]

Hi. I was just looking over your conversation and was wondering, do you still have an example of your previously posted code? It sounds relevant to what I am doing. Could you tell me where I can find it?

Cheers

[aaronw.ca]

If you are referring to the AVTransmit2.java code posted previously be Stephan, you can find it in this thread:

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=IP%20Communications%20and%20Video&topic=IP%20Phone%20Services%20for%20Developers&CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.1dd5d0da/18

Stephan initially posted it at the beginning of that thread, but the download expired, so he reposted it on the second page of the thread.

[Developer22]

Hi I am doing a simillar project- creating an IVR using jtapi and CISCO IP phones, except the CTI route point calls an extension. The phone rings and then when I pick it up, the program outputs that transmission is in progress and the AVreceive is waiting for RTP data. The problem is a) there is no sound over the phone and b) after a few seconds, the phone goes to engaged tone i.e dead noise first then tone.

Since you seem to be extremely knowledgable in this area, I wondered if you could help me. How do I make the CISCO phone "listen"? I am using AVtransmit and receieve -using the same port number etc. I want to use uni-cast although I am running the ap on one computer (also containing cm). Do you know if you have to use media service bind methods?