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Summary:ASTERISK-14617: Registration fails when the packe contains 'transport=udp"
Reporter:Private Name (falves11)Labels:
Date Opened:2009-08-08 06:36:53Date Closed:2011-06-07 14:00:39
Priority:BlockerRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Registration
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:As you can see below, the registration packet from a very popular SIP Phone, the Eyebeam, includes "transport=udp". The Asterisk fails to accept the registration with a message that says "only Unknown transport is accepted". This is wrong. Registration should never be denied if the transport is udp.

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

REGISTER sip:208.78.161.208 SIP/2.0
Via: SIP/2.0/UDP 192.168.25.100:10026;branch=z9hG4bK-d8754z-b010e67dab504d4e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:19544447408@192.168.25.100:10026;rinstance=ef360e02cd8ff721;transport=udp>
To: "Vswitch198"<sip:19544447408@208.78.161.208>
From: "Vswitch198"<sip:19544447408@208.78.161.208>;tag=8b0c6505
Call-ID: MDZmZTM0YzgxOWNmY2RiYjY4NmMwOTg3NjYxNmUwZDg.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: eyeBeam release 1100v stamp 47069
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 64.150.176.78 : 29629 (NAT)                                                    
[Aug  8 07:19:24] ERROR[70924]: chan_sip.c:12412 register_verify: 'UDP' is not a valid transport for '19544447408'. we only use 'UNKNOWN'! ending call.
vswitchabe*CLI>                                                                            
<--- Transmitting (NAT) to 64.150.176.78:29629 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.25.100:10026;branch=z9hG4bK-d8754z-b010e67dab504d4e-1---d8754z-;received=64.150.176.78;rport=29629
From: "Vswitch198"<sip:19544447408@208.78.161.208>;tag=8b0c6505
To: "Vswitch198"<sip:19544447408@208.78.161.208>;tag=as7681df8c
Call-ID: MDZmZTM0YzgxOWNmY2RiYjY4NmMwOTg3NjYxNmUwZDg.
CSeq: 1 REGISTER
Server: Cisco 3845
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Date: Sat, 08 Aug 2009 11:19:24 GMT
Content-Length: 0
Comments:By: Kevin P. Fleming (kpfleming) 2009-08-08 09:27:37

This bug does not exist in any version of Asterisk prior to 1.6.0, or in ABE, since there is no transport parsing prior to those versions (they only supported UDP so any provided transport parameter was ignored).

By: Private Name (falves11) 2009-08-08 09:30:11

Thanks for the clarification.

By: Mark Michelson (mmichelson) 2009-08-10 09:10:31

I cannot reproduce this issue using rev 211346 of 1.6.2. The Polycom phone I use has in its Contact header:

Contact: <sip:666@10.24.19.242;transport=udp>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER"

As you can see, it also includes "transport=udp" in the registration, but Asterisk responds with a 200 OK. So, it's not the presence of the transport parameter which is causing the problem, it must be something else. To help me figure this out, could you post your sip.conf file so that I may try using a similar configuration to reproduce the problem? Thank you.

By: Private Name (falves11) 2009-08-10 11:31:39

The problem is gone in current version 211391. Please close the case.