Build Information:
version n/a I think for feature enhancement?
redhat fedora latest packaged and latest git build:
wireshark 1.10.6 (Git Rev Unknown from unknown)
Copyright 1998-2014 Gerald Combs <[email protected]> and contributors.
This is free software; see the source for copying conditions. There is NO
warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
Compiled (32-bit) with GTK+ 2.24.22, with Cairo 1.12.14, with Pango 1.34.1,
with
GLib 2.36.4, with libpcap, with libz 1.2.7, with POSIX capabilities (Linux),
without libnl, with SMI 0.4.8, with c-ares 1.10.0, with Lua 5.1, without
Python,
with GnuTLS 3.1.20, with Gcrypt 1.5.3, with MIT Kerberos, with GeoIP, with
PortAudio V19-devel (built May 4 2013 13:59:39), with AirPcap.
Running on Linux 3.13.11-100.fc19.i686.PAE, with locale en_GB.UTF-8, with
libpcap version 1.4.0, with libz 1.2.7, GnuTLS 3.1.20, Gcrypt 1.5.3, without
AirPcap.
Built using gcc 4.8.2 20131212 (Red Hat 4.8.2-7).
[james@nebraska wireshark]$ ./wireshark -v
wireshark 1.99.0 (v1.99.0-rc1-357-g40cf54f from master)
Copyright 1998-2014 Gerald Combs <[email protected]> and contributors.
This is free software; see the source for copying conditions. There is NO
warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
Compiled (32-bit) with GTK+ 3.8.8, with Cairo 1.12.14, with Pango 1.34.1, with
GLib 2.36.4, with libpcap, with libz 1.2.7, with POSIX capabilities (Linux),
without libnl, without SMI, with c-ares 1.10.0, with Lua 5.1, with GnuTLS
3.1.20, with Gcrypt 1.5.3, with MIT Kerberos, without GeoIP, without PortAudio,
with AirPcap.
Running on Linux 3.13.11-100.fc19.i686.PAE, with locale en_GB.UTF-8, with
libpcap version 1.4.0, with libz 1.2.7, GnuTLS 3.1.20, Gcrypt 1.5.3, without
AirPcap.
Built using gcc 4.8.2 20131212 (Red Hat 4.8.2-7).
--
It's very useful to see all SIP flows (i.e. flows for REGISTER, OPTIONS and any
SIP messages based on Call-IDs). Currently "VoIP Call" shows flows for SIP
INVITE flows only.
It is a really easy change to make VoIP flows operate on all SIP packets. I
have done so and it works well. But better to make a cleaner change:
* Add a "SIP _Flow" menu option below the "_VoIP Call" menu item.
Make it as a separate menu item so that:
1. It doesn't change VoIP calls feature used currently.
Some users may want INVITE calls only and not clutter of all SIP flows.
2. If there is some problem in non-INVITE flow then already have
workaround
* Fill in status/comment info for the non-call type of flows as appropriate.
In checking was there any work/discussion going in the VoIP call flow area
(grep mailing list and bugs) I didn't find anything much. So I think this is a
useful thing to do.
I did see this bug:
Bug 4905 - VoIP Calls Prepare Filter unreliable
I was already thinking about the Prepare Filter as it's useful when working
with VoIP or SIP flows. There also might be an opportunity to improve how VoIP
call Flow interacts with filters if work is done in this area. 3 things as
mentioned on comment on that bug:
1. prepare filter does work fine (I think? Does it work for everyone?)
2. one minor behaviour to fix: need to clear filter manually before another
filter will fill in
3. it might be nice if VoIP calls/SIP flow applied filters (currently always
operate on all SIP messages I think.).