[Asterisk-Users] ast_rtp_raw_write errors distorting sound on G729 passthrough

[Andres]

Senad Jordanovic wrote:

>Andres wrote:
>  
>
>>Michael Shuler wrote:
>>
>>    
>>
>>>When I use reinvites everything works perfectly (so phoneA<-->phoneB
>>>directly works fine).  When I shut off reinvites
>>>(phoneA<-->asterisk<-->phoneB) I get the following with PhoneA
>>>initiating the call:
>>>
>>>Mar 12 14:43:23 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write:
>>>Difference is 3360, ms is 440 
>>>
>>>
>>>      
>>>
>>This is a delay issue.  Packets are having greater delay than what
>>Asterisk wants.  We had the same problem.  Check this BUG for a
>>possible workaround:
>>http://bugs.digium.com/bug_view_page.php?bug_id=0001195 
>>
>>    
>>
>
>Hi,
>
>I just tried that. Did you have to recompile * in order for it to work?
>
>  
>
Yes, you have to recompile and install again.  You are changing the 
source code.

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-- 
Andres
Network Admin
http://www.telesip.net