Thanks for your help
The version i have of SIPComm is the latest available on the site.
I can now establish bidirectional audio calls between two SIPComm UAs. It is working fine. Sound is bidirectional.
But I put asterisk in the middle, I get the exception :
"net.java.sip.communicator.media.MediaException: Couldn't set any of the tracks to a valid RTP format
The reason why I need Asterisk is to:
1- record calls
2- create conference rooms...
How can I do this? Any clue about the exception above?
ThankYou very much
From: Brian Burch [mailto:Brian@PingToo.com]
Sent: sexta-feira, 7 de Julho de 2006 18:34
To: User Discussion for sip-communicator
Cc: Carlos Alexandre
Subject: Re: SIP Communicator doubts...
On Fri, 7 Jul 2006 16:34:25 +0100, Carlos Alexandre wrote:
I saw your email at the SIPCommunicator fórum, and took the liberty of writing you to ask you a couple of questions.
As you will see, I have posted your question to the SipCommunicator Users mailing list. This is because other people might benefit from the answers.
Hope you don't mind, and that you can help me out.
Im new to SIPCommunicator, and ive been trying to place calls with it.
If you go to the project home page....
... you will see that the "pre-1.0" version supports audio, but the "1.0-draft" does not (it currently supports only instant messaging).
It isn't that obvious, but the page has an FAQ hyperlink that takes you to the documentation wiki - which gives you the roadmap for project development.
There was a thread on the users mailing list recently where Emil and I discussed what did and didn't work with audio on the pre-1.0 project. You could make basic audio calls via proxies, but there were several
frustrating problems with audio. Some were related to JMF, which changed a lot from java 1.3, to 1.4 and then 1.5. Around the time the code was frozen, I had used SipComm enough to prove it more-or-less worked,
but that it wasn't reliable or flexible enough to use with confidence.
1- I can establish calls, but I cant send voice bidireccionally. Is it me, or the project hasn't reached that point yet?
Your comment indicates you are probably using the pre-1.0 version. The newer 1.0-draft has a VERY different architecture and currently does not support audio at all. Uni-directional media streams were a common
symptom with pre-1.0 and rather difficult to diagnose. For example, I never got a 2-way call to the echo test service at Free World Dialup, even though I could successfully call other FWD users with different kinds of
The developers are (I think) close to the point where they will attempt to "semi-port" the audio support from pre-1.0, but I have no feeling for how difficult that task will be.
2- I tried to place calls over an Asterisk system and the same thing happened. Do you know if this is possible?
A colleague and I "played around" with Asterisk about 18 months ago, but it didn't get us very excited. We were using several different softphone products and SipComm wasn't significantly worse than the rest.
Because we were not very interested in SIP to non-SIP calls, we shelved Asterisk and started working with OpenSER, which is just a SIP Proxy and Registrar. Openser worked well with all the SIP softphones we
tested, but as I said above, SipComm wasn't reliable enough to deploy for non-telecoms users... and it hasn't changed since that time.
A lot of problems with media streams can be attributed to firewall issues. We only found one softphone (a year ago) that worked under almost all topologies - the non-free eyeBeam product.
So, to summarise, you probably should not wait for the audio version of SipComm-1.0, but use a very stable softphone (or hardware device such as a Sipura) to develop your server system. Once you have your
server working well, you'll be in a good position to evalute SipComm when it is ready. If you do not need both SIP and analogue PBX functionality in the same server, then you probably don't need Asterisk.
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