[asterisk-bugs] [Asterisk 0014108]: Asterisk not sending out RTP packets

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Dec 18 12:13:51 CST 2008


The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=14108 
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Reported By:                malaiwah
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14108
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
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:             2008-12-18 11:43 CST
Last Modified:              2008-12-18 12:13 CST
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Summary:                    Asterisk not sending out RTP packets
Description: 
I have Kamailio with MediaProxy in front of Asterisk running, asterisk does
not need/nor have to deal with any nat issues. All it is seeing are public
ip addresses (somewhat, because this test environnement runs on amazon
ec2).

I'm debugging SIP packets and everything looks fine though, but Asterisk
is not sending any RTP packet to my RTP proxy. It is receiving packets fine
though. What is stopping Asterisk to send packets to my RTP proxy? Nat=no
in my sip.conf

In additionnal informations, you will find the asterisk console with sip
debug and rtcp debug. Looking at the SIP headers I would interpret that
Asterisk would have to connect to the RTP proxy right away and not wait
until a packet comes in.

* Our Sender:
 SSRC:          824794298
 Sent packets:  0
 Lost packets:  0
 Jitter:        0
 SR-count:      0
 RTT:           0.000000
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---------------------------------------------------------------------- 
 (0096635) file (administrator) - 2008-12-18 12:13
 http://bugs.digium.com/view.php?id=14108#c96635 
---------------------------------------------------------------------- 
You need to try the latest version of 1.4 and attach the console output,
sip debug, and rtp debug as attachments. Knowing the IP addresses of the
various things involved would also help piece this together. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-18 12:13 file           Note Added: 0096635                          
2008-12-18 12:13 file           Status                   new => feedback     
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