[asterisk-bugs] [Asterisk 0011852]: RTCP Read too short
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jan 28 12:49:46 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11852
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Reported By: wlloyd
Assigned To:
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Project: Asterisk
Issue ID: 11852
Category: Core/RTP
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 01-27-2008 09:33 CST
Last Modified: 01-28-2008 12:49 CST
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Summary: RTCP Read too short
Description:
Reopen of issue 10441
Since upgrading to 1.4.17 from 1.2 I get this with one of my clients 100%
of the time. I have a RTP debug saved of a transaction.
Please email me for full log.
-bill
[Jan 27 10:21:56] DEBUG[7398] rtp.c: Got a FRAME_CONTROL (14) frame on
channel S
IP/kmtel-084102e8
[Jan 27 10:21:56] VERBOSE[7398] logger.c: -- Packet2Packet bridging
SIP/64.1
5.69.133-b49145a0 and SIP/kmtel-084102e8
[Jan 27 10:21:56] DEBUG[7398] rtp.c: Got a FRAME_CONTROL (3) frame on
channel SI
P/kmtel-084102e8
[Jan 27 10:21:56] VERBOSE[7398] logger.c: -- Packet2Packet bridging
SIP/64.1
5.69.133-b49145a0 and SIP/kmtel-084102e8
[Jan 27 10:21:56] DEBUG[7398] rtp.c: Got a FRAME_CONTROL (14) frame on
channel S
IP/kmtel-084102e8
[Jan 27 10:21:56] VERBOSE[7398] logger.c: -- Packet2Packet bridging
SIP/64.1
5.69.133-b49145a0 and SIP/kmtel-084102e8
[Jan 27 10:21:58] WARNING[7398] rtp.c: RTCP Read too short
[Jan 27 10:22:01] WARNING[7398] rtp.c: RTCP Read too short
[Jan 27 10:22:07] WARNING[7398] rtp.c: RTCP Read too short
[Jan 27 10:22:14] WARNING[7398] rtp.c: RTCP Read too short
[Jan 27 10:22:17] WARNING[7398] rtp.c: RTCP Read too short
[Jan 27 10:22:23] VERBOSE[7398] logger.c: Sent RTP P2P packet to
64.69.34.139:72
08 (type 00, len 000160)
[Jan 27 10:22:23] VERBOSE[7398] logger.c: Sent RTP P2P packet to
64.15.69.133:16
122 (type 00, len 000160)
lines 61-74
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wlloyd - 01-28-08 12:49
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I had to send the RTP traffic to an 1.2 Asterisk box and then forward it on
from there. Then I have no issues. The client is running voipswitch.
Currently it causes issues with voice quality. The audio quality is very
poor.
When I route through 1.2 instead of 1.4 it cleans up and sounds fine.
Also the cli error messages disappear.
I can do any traces you would like. tcpdump etc.
-bill
Issue History
Date Modified Username Field Change
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01-28-08 12:49 wlloyd Note Added: 0081284
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