[Asterisk-Users] Strange problem with sjphone and 1.2.1

[Chuck Bunn]

Hi,

Rich I stand corrected you are absolutely right - see 
http://www.voip-info.org/wiki-Asterisk+config+sip.conf

The following appears on the page:


    Please note

    * Asterisk does not yet support SIP over TCP. It only supports SIP
      <http://www.voip-info.org/wiki/view/SIP> over UDP.
    * For Grandstream <http://www.voip-info.org/wiki/view/Grandstream>
      phones: set *dtmfmode=info*
    * Asterisk uses the incoming RTP
      <http://www.voip-info.org/wiki/view/RTP> Stream as a timing source
      for sending its outgoing Stream. If the incoming stream is
      interrupted due to silence suppression then musiconhold will be
      choppy. So in conclusion, you cannot use silence suppression.
      *Make sure ALL SIP phones have disabled silence suppression.*
      There is a solution for the silence suppression problem, see bug
      5374 <http://bugs.digium.com/view.php?id=5374> for details.

Thanks


Rich Adamson wrote:

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