This is a real puzzle that has made my other problem determination efforts more
difficult. I have several platform-specific sip softphones as well as a sipura 2100. I use
FWD's extension 613 (echo test) as my primary problem determination tool.
Basically, when the call connects you hear a recorded message for about 20 seconds.
Then, everything you say into your phone ought to be echoed back to your earpiece.
You simply hang up when finished.
I can call this service reliably and successfully with all my sip-phones EXCEPT
sip-communicator. The symptoms are that I hear the ringing tone after making the call.
Next the sip-communicator window is painted with the JMF panel and I hear the
outgoing message (sometimes this is slow and I lose the start of the message).
However, when I start talking I never hear my own voice. The log4j trace shows me that
the inbound and outbound RTM media streams have been successfully opened.
I know, you'll all say it is my problem... but when I use the identical sip-communicator
configuration to call my sipura everything works perfectly. The media streams connect
in both directions and I can talk and listen at either end of the call. I've done a
line-by-line comparison of the two traces and can't find any significant difference. I
haven't yet tried an ethereal trace, but that is mainly from tiredness.
I can call the service from my sipura, registered with a different sip service provider,
and connect a G711u call. It works perfectly. It seems the FWD echo test is freely
available to any caller without registration or authentication.
Oh yes, the symptoms are identical when I run the sip-communicator test outside my
firewall, so it isn't a NAT/Stun issue.
Has anyone else seen (heard) of this kind of problem? Perhaps someone would try
this test from their own sip-communicator and let me know whether it works or not?