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Summary:ASTERISK-04912: [patch] rport stripped from all via headers not just top one
Reporter:Ivan Tikhonov (ivan tikhonov)Labels:
Date Opened:2005-08-27 04:00:13Date Closed:2008-01-15 15:48:05.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) viarport.txt
Description:when replying on message containing:
Via: SIP/2.0/UDP 127.0.0.1:5071;branch=z9hG4bKejloRzRiS0U0NEI2MEQ5MUJBMzQxRTI5RDFENjMzRkI5MjJERDZF
Via: SIP/2.0/UDP 127.0.0.1:5072;received=127.0.0.1;rport=5072;branch=z9hG4bKE44B60D91BA341E29D1D633FB922DD6E

asterisk produce:
Via: SIP/2.0/UDP 127.0.0.1:5071;branch=z9hG4bKejloRzRiS0U0NEI2MEQ5MUJBMzQxRTI5RDFENjMzRkI5MjJERDZF;received=127.0.0.1;rport=5071
Via: SIP/2.0/UDP 127.0.0.1:5072;received=127.0.0.1;branch=z9hG4bKE44B60D91BA341E29D1D633FB922DD6E


there is no rport=5072 in last via. As far as i can see this may lead to broken nat traversal on proxy which sends to asterisk.

Solution:

rport stripping must be surrounded by
if (!copied) {
...
}
in function copy_via_headers


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

INVITE sip:123@localhost SIP/2.0
Record-Route: <sip:127.0.0.1:5071;lr=on>
Via: SIP/2.0/UDP 127.0.0.1:5071;branch=z9hG4bKejloRzRiS0U0NEI2MEQ5MUJBMzQxRTI5RDFENjMzRkI5MjJERDZF
Via: SIP/2.0/UDP 127.0.0.1:5072;received=127.0.0.1;rport=5072;branch=z9hG4bKE44B60D91BA341E29D1D633FB922DD6E
From: <sip:anonymous@localhost;user=phone>;tag=bbb
To: <sip:123@localhost;user=phone>
Call-Id: 0.106392096512
Cseq: 1 INVITE
Content-Length: 0

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5071;branch=z9hG4bKejloRzRiS0U0NEI2MEQ5MUJBMzQxRTI5RDFENjMzRkI5MjJERDZF;received=127.0.0.1;rport=5071
Via: SIP/2.0/UDP 127.0.0.1:5072;received=127.0.0.1;branch=z9hG4bKE44B60D91BA341E29D1D633FB922DD6E
From: <sip:anonymous@localhost;user=phone>;tag=bbb
To: <sip:123@localhost;user=phone>;tag=as771d7a88
Call-ID: 0.106392096512
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:123@172.16.61.154>
Content-Length: 0
Comments:By: Olle Johansson (oej) 2005-09-06 12:16:22

Any idea on why we do strip those rport headers? There is code in there just for that for a reason...
By: Olle Johansson (oej) 2005-09-06 13:14:17

Please test this patch. Disclaimer on file.

Seems like the previous programmer misunderstood the RFCs. Now we always add ;received to the topmost via and only adds rport=port if we get an empty ;rport in the topmost via.
By: Olle Johansson (oej) 2005-09-09 08:26:27

We are waiting for your feedback!!!
By: Ivan Tikhonov (ivan tikhonov) 2005-09-11 01:58:01

Sorry, can't test - already patched my asterisk and it's in production use.

With nat=always it's clearer to always put rport.

But actually there is no need for adding rport at all because asterisk don't forward sip messages and rport info is useless for downstream proxies and client.
By: Michael Jerris (mikej) 2005-09-11 11:51:14

Ivan, can you provide a disclaimed patch?
By: Olle Johansson (oej) 2005-09-12 10:12:28

This patch is confirmed as solving the problem in ASTERISK-5058
By: Kevin P. Fleming (kpfleming) 2005-09-13 21:54:42

Committed to CVS HEAD, thanks!
By: Digium Subversion (svnbot) 2008-01-15 15:48:05.000-0600

Repository: asterisk
Revision: 6582

U   trunk/channels/chan_sip.c

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r6582 | kpfleming | 2008-01-15 15:48:05 -0600 (Tue, 15 Jan 2008) | 2 lines

handle rport/received in Via headers properly (issue ASTERISK-4912)

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http://svn.digium.com/view/asterisk?view=rev&revision=6582