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

[Senad Jordanovic]

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?