[Sipwitch-devel] Public sipwitch server handling calls between extensions behind router

[Steve Murphy]

Hi all

I just wanted to make a note of what I have observed running a SIP Witch 
0.8.4 server on a public internet server,  conducting tests using SIP 
clients on the same network behind a single router box ( with single 
public IP address).  I am using twinkle softphones as well as a snom m3 
( only a single extension/handset unfortunately ).

I have used the snom phone on this network using a providers asterisk 
service without setting a stun server for some time without any problems 
( admittedly this was the only sip client on the network previously )  
and this fact made me conduct initial tests without a stun server set 
for the snom or the twinkle softphones on the local network.

1) Tests without stun server set for clients:
********************************************
1a) If twinkle client calls snom phone , call is established normally , 
and audio is working.

1b) If snom calls twinkle , twinkle rings,  and call can be picked up,  
but twinkle gets stuck with line status message "establishing call, 
please wait" .  Audio is working , but after about 30 secs twinkle hangs 
up the call , printing status message "Line 1: no ACK recieved , call 
will be terminated".  A similiar problem  is discussed at 
http://osdir.com/ml/voip.twinkle/2008-05/msg00025.html.  The snom phone 
does detects the hangup and it's call timer continues until it is 
explicitly hung up.

1c) If one twinkle1 calls twinkle2 , situation is as 1b above ( 
substituting 'twinkle1' for 'snom' and 'twinkle2' for 'twinkle' in 1b).

Observations: The snom phone always seems to bag the default sip port on 
the router box (5060) regardless of which order the phones are powered 
up  and registered. The twinkle phones 5060 ports are forwarded to 
different ports on the router.


2) Tests with  a stun server  set for the clients:
****************************************************
I set all the clients to use stunserver.org
Now all the calls are established normally,  and can be hung-up  
normally by either end BUT the audio does not work! ( silence at both 
ends ).


Further info:

I have started following Michel de Boer's advice to the author of the 
voip.twinkle thread mentioned above,  and collected some network traffic 
on the server and local network during call set up.  What this shows is 
that for tests 1 ( no stun ) at the point the call is picked up,  the 
server begins to generate SIP packets intended for  the failing ( callee 
) client  using it's LOCAL network address ( e.g. 192.168.1.x ) .  Not 
surprising that the client receives no more control packets relating to 
this call!
When stun is set for the clients,  the server never generates packets 
with 192.168.1.x  addresses - but something (as yet undetermined) is 
going wrong with the UDP peer to peer connection between clients.

Is this a known issue?  Does anyone know where the fault is likely to 
originate from ? ( e.g. clients , server or router )

As a complete bodge - I have hacked a copy of twinkle to not hang up the 
call after 30 seconds, which gives me a usable system,  but would much 
rather prefer to fix the underlying problem.

I can press on and find out more about what is happening to  the 
audio/UDP session when the stun servers are setup,  however  all that I 
know of the  details of SIP calling and SIP network traffic has been 
learned while looking at this problem, so it may take a little time.

I suspect everything will work fine as long as I do not call a client on 
the same local network,  but this is harder for me to test, and clients 
on the same network is useful.

Grateful for any thoughts/ advice

Steve