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

[Michael Shuler]

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
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.

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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