[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Rich Adamson]

I don't believe asterisk has any sip "tcp" support. Its all "udp".

------------------------
> Hi,
> 
> Something else I should mention. Sip uses UDP and TCP packets. TCP 
> packets are used if there is congestion on the network. I am unclear 
> about what mechanism causes sip to switch between UDP and TCP but I 
> believe it is controllable - I believe It would be easier to use QOS 
> though. If UDP is used that packets could time out and you would never 
> know it since UDP is dumb and has no packet loss recovery mechanism. 
> What is the topology of your network. Is the Asterisk box and the client 
> on the same backbone and switch?
> 
> Thanks
> 
> Evil Skymarshal wrote:
> 
> > Hi Chuck,
> >
> > 2005/12/17, Chuck Bunn <chuck.bunn at networkdoc.com 
> > <mailto:chuck.bunn at networkdoc.com>>:
> >
> >     What are you codec and dmtfmode settings in sip.conf and in the sip
> >     phone settings.
> >
> >
> > I use gsm.
> >
> >     If you dmtfmode is set to 'inband' and you are using
> >     anything other than ulaw or alaw codec it wont work.
> >
> >
> > I changed the settings and tried:
> > ---cut---
> > exten => 2000,1,Answer()
> > exten => 2000,2,Wait(1)
> > exten => 2000,3,Playback(hello-world)
> > exten => 2000,4,Hangup()
> > ---cut---
> >
> > Same problem. Sometimes it works but most of the times it doesn't.
> >  
> >
> >     Also since your
> >     hear the phone sometimes you may be experiencing QOS issues on your
> >     network.
> >
> >
> > Of course it could be a QOS problem. But should I hear at least something?
> >
> > cu
> >   ES
> >
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