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Summary:ASTERISK-13584: SIP/tcp calls failing with invalid transport error.
Reporter:Jon Webster (jon)Labels:
Date Opened:2009-02-15 17:18:46.000-0600Date Closed:2009-03-09 15:14:08
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:When placing SIP/TCP calls using IAX or the console to a sip/tcp peer I receive a "'UDP' is not a valid transport" error.

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

; sip.conf from a working 1.6.0 built in 2008.
[general]
tcpenable=yes                   ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)


[exchange12]
type=friend
host=66.192.107.236
disallow=all
allow=ulaw
transport=tcp
port=5060
promiscredir=yes
insecure=port,invite
context=exchange12


   -- Executing [7775555@from-astpbx:4] Dial("IAX2/astpbx-10788", "SIP/exchange12/5555") in new stack
[Feb 15 17:48:15] DEBUG[13526]: chan_sip.c:19971 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw)
 == Using SIP RTP CoS mark 5
 == Using SIP VRTP CoS mark 6
[Feb 15 17:48:15] DEBUG[13526]: chan_sip.c:6033 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
[Feb 15 17:48:15] ERROR[13526]: chan_sip.c:4012 create_addr_from_peer: 'UDP' is not a valid transport for 'exchange12'. we only use 'TCP'! ending call.
[Feb 15 17:48:15] DEBUG[13526]: chan_sip.c:20050 sip_request_call: Cant create SIP call - target device not registred
[Feb 15 17:48:15] DEBUG[13526]: chan_sip.c:4675 sip_destroy: Destroying SIP dialog 4c4e839f639e8078346fa0f166839a8b@66.192.107.196
[Feb 15 17:48:15] WARNING[13526]: app_dial.c:1470 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)


Same error when dialing from the console
   -- Executing [7775555@from-astpbx:4] Dial("Console/dsp", "SIP/exchange12/555
5") in new stack
 == Using SIP RTP CoS mark 5
 == Using SIP VRTP CoS mark 6
[Feb 15 18:07:24] ERROR[13679]: chan_sip.c:4012 create_addr_from_peer: 'UDP' is
not a valid transport for 'exchange12'. we only use 'TCP'! ending call.
[Feb 15 18:07:24] WARNING[13679]: app_dial.c:1470 dial_exec_full: Unable to crea
te channel of type 'SIP' (cause 20 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Auto fallthrough, channel 'Console/dsp' status is 'CHANUNAVAIL'
Comments:By: Paul Belanger (pabelanger) 2009-02-15 23:12:57.000-0600

I get this error too.  Try setting:

transport=tcp,udp

and see what happens.
By: Jon Webster (jon) 2009-02-15 23:37:28.000-0600

Awesome, that works. The sample says defaupt is 'udp', so it would make sense for 'tcp' to be a valid option too.
By: Paul Belanger (pabelanger) 2009-02-16 11:57:02.000-0600

No problem.  I was actually going to raise a bug about this, but never got around to it.

From what I gather transport valid values are (tcp,udp or udp.tcp).  But the problem is, my SIP peer does not support UDP.  So if you define transport=tcp, we get the error you listed above.

While defining transport=tcp,udp fixes the issue, it does not make sense for my SIP peer, since UDP is not a valid option.
By: Leif Madsen (lmadsen) 2009-03-04 12:20:37.000-0600

Assigning this issue to file as perhaps it could be a simple fix?
By: Digium Subversion (svnbot) 2009-03-09 15:14:07

Repository: asterisk
Revision: 180718

U   branches/1.6.0/channels/chan_sip.c

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r180718 | file | 2009-03-09 15:14:06 -0500 (Mon, 09 Mar 2009) | 5 lines

Ensure that the new outgoing dialog to a peer is able to set the socket details, even if the default is present.

(closes issue ASTERISK-13584)
Reported by: jon

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http://svn.digium.com/view/asterisk?view=rev&revision=180718