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

[Andres]

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

>Mar 12 14:43:24 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
>is 2400, ms is 320
>Mar 12 14:43:24 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
>is 720, ms is 110
>Mar 12 14:43:25 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
>is 3688, ms is 481
>Mar 12 14:43:25 DEBUG[1209277232]: rtp.c:943 ast_rtp_raw_write: Difference
>is 640, ms is 100
>
>OK, now here is the really interesting part....
>
>When PhoneA calls PhoneB, PhoneA hears hears a jittering type of sound BUT
>PhoneB hears everything crystal clear.
>When PhoneB calls PhoneA, everything works fine......
>
>OK, I'm stumped.
>
>----------------------------------------
>
>Michael Shuler, C.E.O.
>BitWise Systems, Inc.
>1301 W. Pioneer Parkway
>Peoria, IL 61615
>Office: (217) 585-0357
>Cell: (309) 657-6365
>Fax: (309) 213-3500
>E-Mail: mike at bwsys.net
>Customer Service: (877) 976-0711 
>
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>  
>


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