[asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs

[Ricardo Carvalho]

In fact as far as I know, Asterisk stands in the middle of calls, 
breaking one transaction and initiating another to the other side, doing 
the bridge between them... Although good in some cases like permitting 
to start a new transaction to the next hop changing codecs, in my case I 
don't need that feature because I'm using reINVITEs to implement 
session-timer support in the user agent to solve problems of whong 
accounting if power failure or link happens...
Is there any way to disable those breaks in audio stream?

Regards,
Ricardo.





Andreas Sikkema wrote:
>> My Asterisk server is working fine, although every time that 
>> in the middle of
>> any call there is a reinvite, the user hears a glitch. Why is 
>> this happening?
>> How can I solve this problem?
>>     
>
> That's because a REINVITE is generally used to change from one 
> codec to another. For some reason this involves stopping the 
> existing audio, waiting a little while and then starting a new 
> audio stream. 
>
> So far this one of the reasons why I don't like reinvite...
>
>