[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

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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