[asterisk-dev] Re: No complete handling of SIP Via: header?

[Samuel Tardieu]

>>>>> "Johansson" == Johansson Olle E <olle at voop.com> writes:

Johansson> It is indeed a bug. [...] Please open a bug report.

Bug#8524.

I have attached two small fixes to the bug report
(http://bugs.digium.com/view.php?id=8524). Those have been tested
against single- and multi-valued Via: fields and appear to work fine.
Disclaimer is already on file, so feel free to apply the patches.

Other problems (lost communication because of the incorrect "received"
mpcatop,) have disappeared with these fixes.

Example with a multi-valued Via field:

<--- SIP read from 212.27.52.5:5061 --->
INVITE sip:s at 88.191.14.223:5060;transport=udp SIP/2.0
Call-ID: 17031-VE-0009603c-0b0ecedc2 at freephonie.net
Contact: <sip:212.27.52.5:5060>
Content-Type: application/sdp
CSeq: 613059 INVITE
From: <sip:0661183654 at freephonie.net;user=phone>;tag=17031-GJ-0009603d-3fa4d79f1
Max-Forwards: 26
Record-Route: <sip:C=on;n=0 at 212.27.52.5:5061;lr>
To: <sip:0871765972 at 212.27.52.5;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5061;branch=z9hG4bK-R0-00000192-2973f88f,SIP/2.0/UDP 172.17.20.241:5063;emission,SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-59C6-6E92
Allow: UPDATE,REFER
User-Agent: Cirpack/v4.40 (gw_sip)
Content-Length: 172

[...]

<--- Transmitting (no NAT) to 212.27.52.5:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.27.52.5:5061;branch=z9hG4bK-R0-00000192-2973f88f;received=212.27.52.5,SIP/2.0/UDP 172.17.20.241:5063;emission,SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-59C6-6E92
From: <sip:0661183654 at freephonie.net;user=phone>;tag=17031-GJ-0009603d-3fa4d79f1
To: <sip:0871765972 at 212.27.52.5;user=phone>
Call-ID: 17031-VE-0009603c-0b0ecedc2 at freephonie.net
Seq: 613059 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:s at 88.191.14.223>
Content-Length: 0