asterisk users - Sep 2004 - (Resend) Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs

Got no responses to this, but the list seemed to be down for a while, so
here it is again.  Sorry for the extra bandwidth!

        John



Hi, I've been messing with getting SIP working for days now, with
limited success.  I've got Asterisk set up on a remote server with the
echo test.  Please try it out to verify I've got the server working
right:

    sip:robot at nixon.butchwax.com

Running FC1, ThinkPad T22, headset thru the soundcard.  Asterisk is
asterisk-1.0_RC1.  No NAT.  The phones I've tried so far are as follows.

** Linphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
  Sound is fine.  Doesn't seem to pick up anything from the microphone,
though.

** kphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
  Sound is fine.  Picks up sound from the microphone, but the echo-test
repeats it back after passing it through a Mr-Roboto filter.

** tkPhone:
  Sound is fine.  Doesn't seem to be reading from the mic, no traffic
going over the network after the 'demo-echotest' recording finishes. 
The following errors are continuously repeated from tkPhone:

  sent 63426 (3),received 30228 (3);read 615040 write 293120 need
  608000 jitter 38!!!!!!!!!!!!!!!!!!!!!!!!!! Error:
  !!!!!!!!!!!!!!!!!!!!!!!
  Data won't fit within the current RTP packet size

** SJphone:
  Last night:  sound worked fine.  Actually sent sound from the mic,
which came back after about a 5-second delay, but which sounded quite
good.
  Today:  establishes a connection, but absolutely no sound in or out. 
:P

If anyone's interested in my SRPMs, I'd like to know.  The asterisk RPM
builds the zapata, zaptel, and asterisk sources; you must have your
kernel-source rpm installed for it to build the modules against.

Let me know if there's something obvious I'm missing.  Thanks-

        John

John Morris <asterisk@butchwax.com> writes:
> Running FC1, ThinkPad T22, headset thru the soundcard.  Asterisk is
> asterisk-1.0_RC1.  No NAT.  The phones I've tried so far are as
follows.
I've got NetBSD-current on a Thinkpad X31, ear plugs connected to the
built-in sound card, using integral microphone in the laptop.  Running
Asterisk from CVS, freshly updated.  The only soft phone I've tried on
this machine is kphone, which works very well for me.
> ** kphone:  Check out my SRPM at http://www.bigu.org/SRPMS/
>   Sound is fine.  Picks up sound from the microphone, but the echo-test
> repeats it back after passing it through a Mr-Roboto filter.
I have no problems with sound quality in either direction with kphone.
> ** SJphone:
>   Last night:  sound worked fine.  Actually sent sound from the mic,
> which came back after about a 5-second delay, but which sounded quite
> good.
This happens from time to time for me with kphone.  It's outgoing
sound (from kphone to Asterisk) that's being delayed, as far as I can
tell, and yes, it's about 5 seconds.  When this happens, I just hang
up and try again, and everything is fine.
>   Today:  establishes a connection, but absolutely no sound in or out. 
I was plagued with this too, initially, but figured out that it's
usually one of two causes: either you've got codec trouble (which can
be analyzed by invoking Asterisk with -vvv or thereabout), or it's a
firewall problem.  When changing sip.conf to adjust allowed codecs,
remember that you need to restart Asterisk to be sure the change will
"take" -- a 'sip reload' will not always do the right thing.

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145