[Asterisk-Users] Compare to Skype

[Eric "ManxPower" Wieling]

> 
>> One of my user is praising Skype!!!
>>
>> I cannot figure out anymore what I can improve!
>>
>> This users sip show peers  is jumping from 65 msec to 1800 all the 
>> time. Of course his voice quality is like a morse code with dashes or 
>> dots of connection time.
>> The next minute he calls me via Skype and it works fine !!!! What 
>> indicates that there is no fault on his Internet connection!!!
>>
>> He is using his notebook and Xlite, but also tried the snom 360.
>>
>> Any hints?

There are 2 issues here.

1) Asterisk does not have a RTP Jitter Buffer.    RTP is what is used to 
transport audio for SIP (and other protocols).  This means that ANY 
jitter on the SIP Phone -> Asterisk link will cause audio problems.

2) Asterisk times it's outgoing audio based on the incoming audio. 
Therefore, if there is jitter on the SIP Phone -> Asterisk link then 
Asterisk will replicate that jitter on the Asterisk -> SIP Phone direction.

REMEMBER, a jitter buffer only applies on INCOMING audio (from the 
standpoint of the device).

These two issues are the main reason I have not deployed remote SIP 
phones for my clients.

I believe that BOTH of these issues will be fixed in Asterisk 1.4.x, 
which should be released sometime this summer.

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