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Summary:ASTERISK-12136: no native bridging with SIP over TLS enabled
Reporter:Lukas Auer (lukas)Labels:
Date Opened:2008-06-04 02:58:58Date Closed:2011-06-07 14:07:27
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) no-tls_native-bridging.pcap
( 1) sip_debug_log.txt
( 2) tls_no-native-bridging.pcap.pcap
Description:As soon as you enable SIP over TLS, Asterisk does not establish a native bridging between the phones anymore, meaning that the asterisk server stays in the media path and all the traffic flows from phone A to Asterisk and from Asterisk further on to phone B. (see wireshark capture tls_no-native-bridging.pcap)

As soon as you deactivate TLS again, the traffic is routed directly between the two phones. (see wireshark capture no-tls_native-bridging.pcap)

Mailing list: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033330.html



****** STEPS TO REPRODUCE ******

enable TLS on asterisk server and on both snom phones:

sip.conf:
-----------------------------------------
tlsenable=yes
tlscertfile=/etc/asterisk/ssl/asterisk.ssl
tlscafile=/etc/asterisk/ssl/cacert.crt

snom:
-----------------------------------------
Outbound Proxy = 192.168.0.1:5061;transport=tls
whereas 192.168.0.1 is the asterisk server


****** ADDITIONAL INFORMATION ******

environment and tests:
=======================================
Asterisk server: 192.168.0.1
snom phone, ext 42: 192.168.0.42
snom phone, ext 43: 192.168.0.43
---
tested with asterisk 1.6.0-beta9 (tarball) and jpeeler's srtp version (http://svn.digium.com/svn/asterisk/team/jpeeler/srtpRepository, revision 114319)
---
2 snom 300 phones, firmware 7.1.30, identically configured (apart from account info)
---
all phones and server are in 192.168.0.0/24 subnet (no NAT)
---
Insert the line "canreinvite=yes" in sip.conf for both phones
---
the dial command from extensions.conf is very simple, no 't', 'T', 'h', 'H', 'w', 'W' or 'L' arguments:
exten => 42,1,Answer
exten => 42,n,Dial(SIP/42)

exten => 43,1,Answer
exten => 43,n,Dial(SIP/43)
---
When a call between the two phones is being established, asterisk shows the following message on its CLI:
Native bridging SIP/43-082774b8 and SIP/42-082748c8
---
force a common codec and configure phones accordingly:
sip.conf:
disallow=all
allow=alaw    ;or allow=ulaw
Comments:By: Olle Johansson (oej) 2008-07-02 04:43:04

bbryant: You could at least add a comment here when you assign an issue to yourself, so that the report knows that something's going on. Thanks.
By: Leif Madsen (lmadsen) 2008-12-05 09:46:32.000-0600

This issue isn't actually assigned to anyone, so I'm switching the status to Acknowledged.
By: Joshua C. Colp (jcolp) 2009-05-20 10:27:04

After further research and experimentation I have confirmed that this was another victim of the directrtpsetup bug I fixed yesterday. The latest version from SVN works fine, and the next release will also work fine.