Re: [Sofia-sip-devel] Sofia-sip

2014-08-14 Thread Kai Vehmanen
Hi,

On Thu, 14 Aug 2014, Kai Vehmanen wrote:

> As the first step, I'd like to add you Mike as a admin of the sf.net
> project that hosts the current website. You have by far most merged

this is done now, Mike added as an admin.

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Re: [Sofia-sip-devel] Sofia-sip

2014-08-13 Thread Kai Vehmanen
Hi,

On Mon, 11 Aug 2014, Michael Jerris wrote:

> Kai!  long time no talk.  I'm fine with whatever folks want, just need a 
> little help as far as manpower to clean up what we have back to 
> something usable outside freeswitch.

sorry for the long silence. I've been kind of hoping somebody from the 
original team could have picked up and continued the coordination work, 
but that does not seem to happen (at least I can state I don't have the 
time and possibility to take this role), so indeed time to arrange for 
something else.

As the first step, I'd like to add you Mike as a admin of the sf.net 
project that hosts the current website. You have by far most merged 
patches in current codebase out of those still actively participating on 
the list. Even if you don't take over directly, you have the possibility 
to pass the project onwards (or at least add a redirect to the FreeSWITCH 
branch).

I understand there are concerns with moving completely to FreeSWITCH for 
those who've been using sofia-sip for other purposes, but there is quite a 
lot involved in running a full independent project. With FreeSWITCH, the 
codebase could be maintained with potentially less overhead, and in 
exchange for making the Sofia-SIP instance in FreeSWITCH tree more 
generally usable (if anything is needed to begin with to that end), 
FreeSWITCH infra could be used to review and merge patches to a single 
trunk (from the many gitorious trees that exist now). So potentially 
win-win for all involved.

But yeah, maybe I shouldn't intervene/propose anything beyond this, 
it's upto you who continue to decide.

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Re: [Sofia-sip-devel] Fwd: Sofia-sip

2014-08-10 Thread Kai Vehmanen
Hi,

[resend to list]

On Sat, 9 Aug 2014, Nikos Balkanas wrote:

> Apparently kaiv is a sofia-sip active administrator, according to 
> sourceforge ops. Do they mean list moderator?
> Do you know him?

I'm still here. :)

I still have admin rights to both the SF and gitorious projects (except 
for the SF net mailing list, only Pekka has the admin password). If there 
is interest to take the project over, I can help with the transition. 
Ideally I'd like to get an ack from Pekka (and Mike), but let's see if I 
can reach him still.

For all practical purposes, the version of Sofia-SIP in FreeSWITCH trunk 
is the new upstream for the project, but I agree updating the old website 
to reflect this would be appropriate (and avoid confusion).

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Re: [Sofia-sip-devel] SOA an Media Session

2012-02-17 Thread Kai Vehmanen
Hi,

On Mon, 6 Feb 2012, abue.sofiasip.dev wrote:

> thanks for your fast feedback! I already check SofSipCli, but especially
> when handling audio AND video it does not suite the purpose, besides it
> has a lot of specific assumption inside the code when talking about
> SDP/Codec negotiation (e.g. fixed to PCMU).

yes, it's just a as-simple-as-possible example how to get a call going.

> I will check telepathy-rakia for further studies - thanks for this hint!
>
> What about functions like soa_generate_offer() or soa_process_answer() -
> are they meant to be used by the application (ua) or just by other parts
> of Sofia Lib? Does the user agent really have to handle stuff like
> "parsing of incoming SDP" and "negotiate codecs of media subsystem upon

No, SofiaSIP takes care of the negotiation, but you need to provide what 
media, and what codecs, your media implementation supports. A very 
complete example is in Telepathy-Rakia, e.g.:

http://cgit.collabora.com/git/user/cassidy/telepathy-rakia/tree/rakia/media-session.c

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Re: [Sofia-sip-devel] SOA an Media Session

2012-02-02 Thread Kai Vehmanen
Hi,

On Thu, 2 Feb 2012, abue.sofiasip.dev wrote:

> I'm are trying to implement an user agent using Sofia-SIP library and
> GStreamer as media stack. It's important that the application is able to
> run on embedded devices, like beagleboard.

telepathy-rakia (formerly known as telepathy-sofiasip) is open-source, 
uses sofia-sip and gstreamer, and is used for SIP VoIP calls e.g. on Nokia 
N900 and Nokia N9 (which btw have the same CPU as beagleboard, so fairly 
perfect match):

o http://telepathy.freedesktop.org/wiki/Components

One very compact example is sofsip-cli, but this is so outdated that 
probably not a good example anymore:

o http://gitorious.org/sofia-sip/pages/SofSipCli

Br,
Kai

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Re: [Sofia-sip-devel] soa example

2007-11-09 Thread Kai Vehmanen
Hi,

On Thu, 8 Nov 2007, Simon Perreault wrote:

> I've looked at every single application mentioned on
> http://wiki.opensource.nokia.com/projects/SofiaApplications. I grepped for
> any kind of usage of soa.
>
> Nobody uses it.

most/all apps use 'soa' through 'nua' interface. See for instance:

http://sofia-sip.org/repos/sofsip-cli/src/ssc_sip.c

... and grep for "SDP O/A note" and "SOA".

If you want to use 'soa' directly, I recommend taking look at 'nua' 
sources in sofia-sip.

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Re: [Sofia-sip-devel] 1.12.6 Application Interface

2007-10-03 Thread Kai Vehmanen
Hi,

ok, forget my reply yesterday. :)

On Wed, 3 Oct 2007, Fabio Margarido wrote:

>> 1.12.4 allowed you to create the su_root object in one thread, and then
>> run it in another. This was changed in 1.12.6. But even with 1.12.4, you
>> weren't supposed to access nua objects from two threads at the same time!
>> This hasn't changed since 1.12.4 AFAIK.
>
> If that's so, I believe nua documentation is a bit misleading. Quoting
> the section 'Initialization and deinitialization':
>
> "If the application is not just responding to incoming SIP messages
> there must also be means to send messages to NUA. This can be handled
> for example by having a separate thread that calls NUA functions to

So it would seem this is supported as well after all and the documentation 
is in fact correct. Basicly the public nua functions implement the 
approach I outlined in my previous post, except using a different wake-up 
mechanism. If I'm now reading the code right, the request execution 
worst-case delay will be somewhat worse when submitting nua requests from 
other threads, vs implementing the local pipe trick I described in my 
mail. But I'll have to double-check this...

Oh well, I've never used the API this way myself, and didn't bother to 
recheck the sources before pushing 'send' yesterday. Sorry for adding to 
the confusion, and thanks Fabio for pointing this out!

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[Sofia-sip-devel] FYI: mailing list and website for the NICE project (open-source IETF ICE implementation)

2007-10-03 Thread Kai Vehmanen
Hello all,

a mailing list, and a dedidicated website, have been launched
for the NICE project. If you are interested in joining the effort, 
please check out the following:

http://lists.freedesktop.org/mailman/listinfo/nice
http://nice.freedesktop.org/wiki/

We are currently focused on merging the fullmode branch
to the mainline, and updating the fullmode code to 
latest -18 ID of ICE. While waiting for the merge to
complete, we have now provided snapshot tarballs of
the fullmode version (i.e. version of the codebase implementing
the latest IETF spec) -- links are in the wiki.

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Re: [Sofia-sip-devel] 1.12.6 Application Interface

2007-10-02 Thread Kai Vehmanen
Hi,

On Tue, 2 Oct 2007, Jerry Richards wrote:

> Currently, with 1.12.4, my call control application thread handles messages
> from other application threads that include removing/adding new lines (i.e.
> multiple NUAs).  If it is now a 1.12.6 requirement that the su_root_ thread
> calls nua_create(), how would my call control thread communicate to the
> su_root_ thread to later invoke nua_create()?
[...]
> I don't understand this new 1.12.6 design.  My current application design is
> the result of guidance received via this mailing list.

I haven't followed this thread from the start, so my apologies if I've 
missed something, but there seems to have been some confusion:

Sofia-SIP is an event-driven library where you can have multiple event 
loops, but each of those has to be accessed from at most one thread.

So if you have a multithreaded application, you can:
   - run multiple separate nua instance (with their own su_root), one per
 thread, or
   - have multiple threads, but concentrate all access to nua objects
 (sharing a single su_root) to a single thread

Now with your current design, su_root_run() might trigger calls to nua 
code (from timers, network i/o, etc), while you could be at the same time 
calling nua functions from other threads. This leads to potentially 
entering the event loop in multiple places, and this is not supported (and 
can lead to hard to debug errors).

1.12.4 allowed you to create the su_root object in one thread, and then 
run it in another. This was changed in 1.12.6. But even with 1.12.4, you 
weren't supposed to access nua objects from two threads at the same time! 
This hasn't changed since 1.12.4 AFAIK.

> Perhaps there is a patch for 1.12.6 so that I can preserve my application as
> shown below?  That is, I invoke nua_create/nua_destroy from my main
> application thread, and call su_root_run() from a separate thread, as shown
> below.

This approach breaks the event loop model and can lead to errors (as your 
code not only creates the nua object, but also accesses it from multiple 
threads at the same time). What you could do instead (one approach):

- have one thread communicate with the sofia nua API (if multiple nua
   objects, use a single su_root for all ->single event loop)
- provide a mutex protected API to communicate with this thread from
   other parts of your application
- use a local pipe (registered to su_root()) to wake the
   sofia-sip thread from su_root_run() when you have to handle
   requests from your other threads
 - once all requests are processed, continue running su_root_run()
   in the sofia-sip thread
- you can freely submit requests to Sofia-SIP from any thread of
   your program

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Re: [Sofia-sip-devel] Problems with autogen.cmd on Windows XP

2007-07-20 Thread Kai Vehmanen
Hi,

On Thu, 19 Jul 2007, Pedro Sanchez wrote:

> Hmm, total silence ... I'm still stuck here. Has anyone compiled
> sofia-sip under Windows recently? Which version did you use?'

ok, I now found out the problem. I've had no problems with gawk 3.1.5, but 
it seems gawk 3.1.0 is too old. And unfortunately only 3.1.0 is available 
at unxutils.sourceforge.net (they used to have newer releases, but as you 
pointed out, those links are dead now).

However, I now tested with mawk 1.3.3 and that works as well. I tested two 
different binaries, and both worked:

   - http://gnuwin32.sourceforge.net/packages/mawk.htm
 mawk.exe md5sum 4d45212b22694d42e7bb7dee93c9f23a
   - http://ftp.uni-kl.de/pub/windows/cygwin/release/gawk/
 gawk.exe md5sum d984ab0c897a48d82dc657759842b3ec
   - http://klabaster.com/freeware.htm (binary only)
 mawk.exe md5sum 1565b1a0fcb3bae80dad0a559815ccba

PS I'll update the sofia-sip/win32/README.txt with the above.

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Re: [Sofia-sip-devel] open-source ICE

2007-07-18 Thread Kai Vehmanen
Hi,

On Mon, 9 Jul 2007, Kai Vehmanen wrote:

> somewhat related, we have been working on a patch to add full IETF ICE 
> support to the Freedesktop NICE project. The project page is at:

as a follow-up, I now added a test media implementation using NICE to 
sofsip-cli. Here's a quick intro how to run automated tests with this 
code:

[getting and compiling the software]
sh> darcs get http://projects.collabora.co.uk/~kvehmane/nice/
sh> cd nice
sh> ./configure && make && make install
sh> darcs get http://sofia-sip.org/repos/sofsip-cli
sh> cd sofsip-cli
sh> ./configure && make
sh> cd src

[server client]
sh> ./sofsip-cli sip:[EMAIL PROTECTED] --media-impl=nice --autoanswer
# register to network with sofsip-cli
sofsip-cli> r
# if password protected account, provide the password
sofsip-cli> k mypassword

[test client]
sh> ./nice_tester sip:[EMAIL PROTECTED] sip:[EMAIL PROTECTED] testpassword && 
echo "SUCCESS"

This allows for fairly automated testing from different access networks. 
Do let me know if you have success (or even problems) with this. It would 
be interesting to know how current NICE works in different networks.

The 'nice_tester' program is also a good example how to write simple SIP 
test apps with Sofia-SIP.

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Re: [Sofia-sip-devel] Need gawk.exe for Windows build

2007-07-18 Thread Kai Vehmanen
Hi, 

On 18 July 2007, Pedro Sanchez wrote:
>I want to compile sofia-sip under Windows XP and according to 
>the documentation I need the executable gawk.exe. This file is 
>supposed to be available from http://unxutils.sourceforge.net 
>but the download links are broken.

the project's sourceforge.net download page is still working:
http://sourceforge.net/project/showfiles.php?group_id=9328

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[Sofia-sip-devel] fyi: a new SIP client using sofia-sip

2007-07-16 Thread Kai Vehmanen
Hi,

I'm pleased to point you to a beta release that adds SIP support to the 
Maemo platform, announced today at GUADEC2007. As you can see from the 
package list, Sofia-SIP is of course used as the SIP stack! :)

Please see the full announcement sent to maemo-developers list:
- http://article.gmane.org/gmane.comp.handhelds.maemo.devel/10833

To get the beta release, you can go directly to:
- http://rtcomm.garage.maemo.org/

Btw; if you have an application/device using sofia-sip that hasn't
  been mentioned on the list yet, feel free to send a brief
  mail about it here!

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Re: [Sofia-sip-devel] Sofia-sip TLS under windows

2007-07-14 Thread Kai Vehmanen
Hi,

On Wed, 11 Jul 2007, chen.liang.sip wrote:

> I want to know whether sofia-sip 1.12.6 has already supported TLS under 
> windows since i've found the same question in mail list, or it justs 
> works under linux depending on OpenSSL ?

I'm afraid situation hasn't change since the last discussion, see:
http://osdir.com/ml/telephony.sofia-sip.devel/2006-11/msg00044.html

So in theory it should work, but nobody has done the work of integrating 
it to the win32 project files (plus probably some modifications to 
the source code to make it compile).

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Re: [Sofia-sip-devel] building sofia sip 1.12.6 on windows

2007-07-13 Thread Kai Vehmanen
Hi,

On Fri, 13 Jul 2007, aviad rozenhek wrote:

> Actually I am using msvc 2005 proffessional.
> I have the vista SDK installed.

then it should work (ws2tcpip.h should define IPPROTO_IPV6). :P

Anyways, I'm currently updating the build rules a bit (plus the win32 
README.txt), and will make it possible to build without IPv6 on win32. 
Maybe we should actually make that the default to minimize build issues.

You could try the following:

win32/su_configure.h
--cut--
-#define SU_HAVE_IN6 1
+#define SU_HAVE_IN6 0
--cut--

libsofia-sip-ua/su/sofia-sip/su.h
--cut--
-#  elif SU_HAVE_IN6
-#include  /* From "IPv6 Tech Preview"  */
-#  else
-#error Winsock with IPv6 support required
--cut--

Btw; with W2k and older platform SDKs, you need the IPv6-tech-preview SDK 
even if you are not using IPv6. These older platform APIs are missing many 
socket API features that sofia-sip is using.

> is the win32 build something people use or is it usually untested?

Yes, I know at least one commercial product using Sofia-SIP on win32. Also 
Freeswitch (which uses sofia-sip) supports win32.

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Re: [Sofia-sip-devel] building sofia sip 1.12.6 on windows

2007-07-12 Thread Kai Vehmanen
Hi,

On Wed, 11 Jul 2007, aviad rozenhek wrote:

> I am trying to compile release sofia-sip-1.12.6 on windows.
[...]
> 1>d:\samples\sofia-sip-1.12.6\sofia-sip-1.12.6\libsofia-sip-ua\su\sofia-sip\su.h(73)
> : fatal error C1083: Cannot open include file: 'tpipv6.h': No such
> file or directory

you need the IPv6-preview APIs installed, see sofia-sip\win32\README.txt 
for more details.

Alternatively you can disable IPv6 altogether by modifying 
sofia-sip\win32\sofia-sip\su_configure.sh (set SU_HAVE_IN6 to zero).

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Re: [Sofia-sip-devel] Incoming Re-INVITE SDP Content

2007-07-10 Thread Kai Vehmanen
Hi,

On Tue, 10 Jul 2007, Jerry Richards wrote:

> So does this mean there is a bug in the sofia-sip SOA engine (v. 1.12.4)?
>
> In other words, it should match the following two codecs as the same?
>
> ... (a): rtpmap:0 PCMU/8000   <---EQUAL--->   ... (a): rtpmap:0 G711/8000

well, it depends. :)

According to specs, that's not a bug as PCMU and G711 are not the same 
thing (and I'd say the client that puts G711 there definitely has a bug,
see RFC3551).

But, but, as the world is not perfect, maybe the stack should just compare 
the payload numbers in these cases. The approach described by Arsen seems 
sane to me...

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[Sofia-sip-devel] open-source ICE (was: Re: changing sofia sip to use libevent?)

2007-07-09 Thread Kai Vehmanen
Hi,

On Fri, 29 Jun 2007, wensong zhang wrote:

> We want to use sofia sip stack with our modules such as STUN, TURN and ICE.
> Those modues use libevent library for event handling. We still like to keep
> single-thread event-driven architecture.

somewhat related, we have been working on a patch to add full IETF ICE 
support to the Freedesktop NICE project. The project page is at:

  http://nice.freedesktop.org/wiki

That code implements a fairly old ICE, but our patch adds support to the 
latest -16 draft, plus adds full-mode support. The patch is currently in 
the merge queue waiting for review:

  http://projects.collabora.co.uk/~monkey/nice-fullmode/

You can get the full patched source (darcs) tree with:

  sh> darcs get http://projects.collabora.co.uk/~kvehmane/nice/

PS Sorry for the off-topic marketing, but I thought this might be
interesting to some users of sofia-sip...

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Re: [Sofia-sip-devel] re-subscribe, re-publish and re-register fail after some time

2007-06-16 Thread Kai Vehmanen
Hi,

On Fri, 15 Jun 2007, Jan Van den bosch wrote:

> I left my NUA application on during the night, and when I returned, my
> console showed this:
>
> tport_vsend(0x43434e0): Invalid argument with (s=12 */172.16.58.69:5060)
> nta: SUBSCRIBE (84798893): Invalid argument (22) with */[172.16.58.69]:5060

that could happen if the network interface the NUA instance was bound to 
was shut down (and maybe restarted later).

Sofia-SIP provides a mechanism to monitor these and automatically 
restart: NUTAG_DETECT_NETWORK_UPDATES() which takes one of 
NUA_NW_DETECT_NOTHING, NUA_NW_DETECT_ONLY_INFO, NUA_NW_DETECT_TRY_FULL. 
But unfortunately the low-level implementation is only available for Mac 
OS X at the moment (libsofia-sip-ua/su/su_os_nw.c).

An alternative way is to monitor for network changes in the client (for 
instance using Linux netlink APIs), and restart NUA if there are changes 
in the local network interfaces.

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[Sofia-sip-devel] ANN: sofsip-cli 0.13 released!

2007-06-04 Thread Kai Vehmanen
Hello all,

the devel tree has gathered some small changes, so it's probably a 
good time for another release.

- tarball at:
   
https://sourceforge.net/project/showfiles.php?group_id=143636&package_id=179933
- signatures
   md5: 0742b6b214dd29e9a02761f5957c3b6e sofsip-cli-0.13.tar.gz

Changes in release: 0.13 (2007-05-31)
-

- Use a random local UDP port for SIP by default (can be overridden
   with the "--contact" command-line option).
- Fixed bugs in initiating multiple media sessions with "gstreamer"
   backend.
- Fixed bug in responding to an 401/407 authentication challenge with
   no realm.
- Request for a lower, 180sec, registration refresh interval from
   the registrar by default.

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Re: [Sofia-sip-devel] working stun - symmetric response routing anyone ?

2007-05-31 Thread Kai Vehmanen
Hi Marcus and others,

On Tue, 29 May 2007, Marcus Priesch wrote:

>> it should work out of the box, but requires that server supports the rport
>> (RFC3581) extension.
>
> so the "Yes, the above API does not work at the moment. :(" comment from
> you in your previous mail wasn't meant for the rfc3581 part ?

no, that was specifically about using STUN binding discovery to generate 
the contact/via for REGISTER (which is not standardized anywhere but still 
implemented in many clients).

> unfortunately i haven't read the rfc before i was posting here :( -
> shame on me !

No, no problem. The fact that we have some code for STUN binding discovery 
in our codebase proves that we haven't outright given up on the idea. It's
just that more recent stuff - like the IETF sip-outbound draft - is 
looking much more promising to solve this specific issue.

> instead i got my info from a book over Voip - which was a bit unclear in
> this respect! furthermore packet analyzing ekiga's efforts also showed
> that when using nat, the IP address in the Via: header is actually the
> public address.

RFC3581 is just a much better way to solve this.

> the point why i came to this is caused by the fact that my asterisk
> (1.2.17) doesn't honour the "received" line correctly in the Via header.
>
> as far as i understand it:
> - asterisk should insert the received= header when the ip in the Via
>   and the originating ip of the packet differ
> - asterisk should fill the rport= header with the actual port of the
>   originating packet
> - asterisk should perform the actions required by this packet
> - when sending the answer back it should honor the received= and rport=
>   headers if they are available ...

Yes, it should basicly send the packet back to the IP:port it received
the request from (for datagram transports). The problem with "fixing up" 
Via/Contact with STUN is that you might discover the wrong IP (known 
limitation of STUN/RFC3489).

For a good overview of state-of-art solutions to these issues can be found 
in the following IETF document (work in progress):

http://tools.ietf.org/wg/sipping/draft-ietf-sipping-nat-scenarios/

> interestingly it simply sends the response back to 192.168.0.6 - which
> isn't reachable because its natted ;). in the sip.conf i have enabled
> rfc3581 for nat (also sip show users, sip show defaults show me this).

Hmm, sounds like a bug...

> only if i tell asterisk that my client is fully natted (nat=yes) it
> sends the response back to my nat firewall (but in this case, the doc
> says its now ignoring any settings found in SIP/SDP and blindly sends
> back to the sender - that was the reason why i thought the outgoing
> REGISTER message was buggy - and the fact that ekiga's (natted) REGISTER
> also looks different - and works !).

Yes, this is what should happen.

> i think that setting nat=yes in asterisk so that it ignores rfc3581 and
> received= headers is not the way to test this ... what do you use as a
> "reference registrar" ?!?!

I'm using some proprietary servers for testing, but for instance 
SER/OpenSER does the right thing with Sofia-SIP's REGISTERs. Freeswitch 
should work as well, but that's kind of expected (check the SIP stack that 
is used). ;) Maybe I should try with a recent Asterisk as well.

>> new sockets. It would seem the current code does not do this, which
>> suggests major refactoring is needed.
>
> the problem is that when it binds to 127.0.0.1 it fails and doesn't try
> to bind to e.g. 192.168.0.6 (the "real" ip-address) ... and this is a
> major problem when the nua wants to initialise the STUN transport - so i
> think this is currently broken ... although not needed by anyone ;)

Yeah, this should still be fixed. Even though RFC3581 is a better (and 
using TCP instead of UDP is even better!) solution, in some special 
networks fixing Via+Contact with STUN still makes sense, so it wouldn't 
hurt to support it.

> hmmm ... maybe some clarification on the use of STUNTAG_SERVER and
> NUTAG_OUTBOUND tags - from the users point of view would be helpful ;)

... yes, NUTAG_OUTBOUND()'s documentation was improved quite a bit 
recently, but there's still room for improvement. Maybe we should add a 
link to this thread to STUNTAG_SERVER's documentation. ;)

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Re: [Sofia-sip-devel] working stun - symmetric response routing anyone ?

2007-05-26 Thread Kai Vehmanen
Hi,

On Tue, 22 May 2007, Marcus Priesch wrote:

> thanks for the reply, sad enough to hear that stun doesn't work for the
> signalling :( but hey, nobody is perfect ;) - so, whats the way to go
> then ? - how do i manage to get the rport stuff working correctly ... do
> i need to do some manual tuning, or should it work out-of-the-box ? the

it should work out of the box, but requires that server supports the rport 
(RFC3581) extension.

> webpage says that "symmetric response routing" is supported ...

Yes, that refers to RFC3581 - "An Extension to the Session Initiation 
Protocol (SIP) for Symmetric Response Routing".

> and i get another transport determined by stun module - i think this
> should be fixed in nta_agent_add_tport (i think when it wants to bind
> the server ?!? - it should not try the 127.0.0.1 transport - in general:
> for what reason do you bind to 127.0.0.1 actually ? who needs it ?)

No need. Now I'm not personally familiar with this code, but in my view it 
should perform a STUN query from all bound UDP sockets, and not create any 
new sockets. It would seem the current code does not do this, which 
suggests major refactoring is needed.

But then I'm still unsure why anyone really needs STUN for signaling? I 
personally think it's a very dangerous idea, because with NATs that do 
endpoint dependent mapping (allocate the public side IP:port based on 
destination), STUN would discover an invalid public contact (the mappings 
will be different for STUN and SIP servers). This is bad as the 
registration is succesful, but in reality it can be that nobody can make 
calls or send messages towards to you. And as a client you have no way to 
figure out that this is the case. This is very, very, very bad.

Use of rport/3581 is much safer as the discovery is done against the same 
destination as the actual REGISTER message is sent to. Thus this discovery 
works even if you have a endpoint-dependend-mapping NAT.

In short, STUN for signaling is only useful in controlled SIP services, 
where both the SIP servers and access networks (NATs) are provider 
controlled. If you have this much control, you might just as well 
configure your service to enable rport/3501.

... of course, if someone can prove me wrong, I'm willing to change my 
mind and consider a serious look at the STUN code in nua/nta. :)

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Re: [Sofia-sip-devel] Handling of SDP in UA Application

2007-05-26 Thread Kai Vehmanen
On Mon, 21 May 2007, Jerry Richards wrote:

> Is there an example of how an NUA should handle the SDP body?  I realize

Pekka already replied to this, but an additional source code link:

- http://sofia-sip.org/repos/sofsip-cli/src/ssc_media_fsgst.c

In the above we implemented the steps Pekka describes in his mail.
Some of the code interfacing with Sofia-SIP's sdp module is in a separate 
file:

- http://sofia-sip.org/repos/sofsip-cli/src/sdp_utils.c

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Re: [Sofia-sip-devel] T.38 Fax Reinvite.

2007-05-26 Thread Kai Vehmanen
Hi,

On Fri, 25 May 2007, Massimo Cetra wrote:

> This basically allowme to do something like:
>
>   nua_invite(tech_pvt->nh,
>   SOATAG_USER_SDP_FORCE( 1 ),
>   SOATAG_USER_SDP_STR( buf_final ),
>   TAG_END()
>   );

yes, so this is exactly what I was thinking about (see previous mail). :)

> There is another problem though.
> Even if i am able to send a reinvite with the content i like, i absolutely 
> cannot handle a received T38 reinvite.

To disable automatic responses to incoming reinvites, pass 
NUTAG_AUTOANSWER(0) to nua_invite/nua_respond(). See...

  - 
http://sofia-sip.sourceforge.net/refdocs/nua/nua__tag_8h.html#8d92bee715bb6056c93a180f4c8fc2b7

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Re: [Sofia-sip-devel] T.38 Fax Reinvite.

2007-05-26 Thread Kai Vehmanen
Hi,

On Fri, 25 May 2007, Pekka Pessi wrote:

>> I propose we add a tag that can be passed in on revinvite that tells it
>> to not merge in previous m= lines... and to only add those are in the
>
> That is probably what is needed. I had a peek on soa_static.c, there
> might be some problems because we do not keep old user sdp around..

hmm, what if the interface (nua_respond()) would require you to pass a new 
user-sdp, plus a boolean tag indicating that you wish to modify media 
session (including media types)?

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Re: [Sofia-sip-devel] working stun - symmetric response routing anyone ?

2007-05-21 Thread Kai Vehmanen
Hi,

On Wed, 16 May 2007, Marcus Priesch wrote:

> first of all, thanks for that great piece of software, i am using it now
> for quite a while and it's really nice ;) - under python via ctypes -
> YEAH !

thanks, that's good to hear! :)

> but i im totally stuck with the stun thing - even the sofsip-cli isn't
> working as expected.
>
> i have searched the mailing list, looked at code which is using
> sofia-sip (sofsip-cli, tapioca and freeswitch) but no success.
>
> using the stun module standalone works perfect ! - even in python ;)

Yes, this is a known issue I'm afraid. STUN is not currently enabled for 
signaling (i.e. for updating the "Contact" for registrations). It _is_ 
usable for a SIP client wishing to use sofia-sip's STUN module to discover 
IP addresses for media (i.e. to SDP).

This has not been an urgent item for us at least, as use of "rport" 
(RFC3581) gets you basicly the same results (for SIP signaling). And in 
the future, the 'SIP outbound' extension will completely remove the need 
for clients to guess their public Contact bindings. There are however many 
people interested in this, so patches are very much welcome.

>self.ss.nua_register \
>( self.register_h
>, self.ss.nutag_registrar, ctypes.c_char_p ("my.own.regist.rar")
>, self.ss.stuntag_server, ctypes.c_char_p ("stun.voipbuster.com")
>, self.ss.nutag_outbound, ctypes.c_char_p ("natify")
>, self.ss.nutag_outbound, ctypes.c_char_p ("use-rport")
>, None, None
>)

Yes, the above API does not work at the moment. :(

Lot of the code is there already, but someone just has to finish the work 
in order to make it really work.

> stun_handle_init: entering.
> stun_handle_init("stun.voipbuster.com"): called
> stun_atoaddr: entering.
> stun_bind: entering.
> stun_discovery_create: entering.
> assign_socket: entering.
> stun_request_create: entering.
> stun_make_binding_req: entering.
> stun_send_binding_request: entering.
> stun_send_message: message sent to 194.221.62.209:3478

Hmm, this goes further than I remembered.

> tport_listen(0x816c5a8): stun_bind()(pf=2 UDP/[127.0.0.1]:32861): Invalid 
> argument

Ugh, this looks like a bug. Hmm patches are welcome! :)

> read something on the list that 1.12.3 had problems with stun, and
> 1.12.2 was fine ... but the same there ... hmmm.

Yes, this applied to the STUN module itself, but 1.12.6 and the current 
devel tree should be fine (you can test with the 'stunc' test tool).

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Re: [Sofia-sip-devel] Obtaining REALM from WWW-Authenticate contained in 407 response to REGISTER message.

2007-04-18 Thread Kai Vehmanen
Hi,

On Wed, 18 Apr 2007, Matthew O Connor wrote:

> I am having trouble figuring out how to get access to the realm contained in 
> the
> WWW-Authenticate of the 407 response to a REGISTER message I send to a PBX.
>
> The contents of the WWW-Authenticate are shown below.
>
> WWW-Authenticate: Digest algorithm=MD5, realm="openpbx.org", nonce="0aab13c6"

an example of how this should work is at (although the gaim sofia plugin 
should work as well, but just in case some case is not handled):

- http://sofia-sip.org/repos/sofsip-cli/src/ssc_sip.c

See ssc_store_pending_auth(). In short you should get the realm by doing 
the following steps:

step 1:
   sip_www_authenticate_t const *wa = sip->sip_www_authenticate;
or
   sip_proxy_authenticate_t const *pa = sip->sip_proxy_authenticate;
or
   tl_gets(tags,
  SIPTAG_WWW_AUTHENTICATE_REF(wa),
  SIPTAG_PROXY_AUTHENTICATE_REF(pa),
  TAG_NULL());
step 2:
   msg_param_t const *au_params = wa->au_params;
or
   msg_param_t const *au_params = pa->au_params;
step 3:
   const char *realm = msg_params_find(au_params, "realm=");

In other words either you pick the realm from the SIP headers (the common 
case), or alternatively the stack provides the challenge in tags (a rare 
case, can't come up with a practical example of where this is used :)).

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Re: [Sofia-sip-devel] Issue with glib_root

2007-04-04 Thread Kai Vehmanen
Hi,

On Wed, 4 Apr 2007, Enrico Marocco wrote:

> it seems that a root objects created by su_glib_root_create() have some
> issues with timers.  I haven't had time for investigating the problem,

yes, this was fixed in the 1.12.5 release - see the fixed bugs list: 
http://sofia-sip.sourceforge.net/relnotes/relnotes-sofia-sip-1.12.5.txt

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Re: [Sofia-sip-devel] Compiling sofia-sip

2007-03-15 Thread Kai Vehmanen

Hi,

On Thu, 15 Mar 2007, Deniz Koçak wrote:


_su_wsaevent_port_create
su_port.obj : error LNK2001: unresolved external symbol
_su_wsaevent_clone_start
Debug/libsofia_sip_ua.dll : fatal error LNK1120: 2 unresolved externals
Error executing link.exe.

libsofia_sip_ua.dll - 3 error(s), 0 warning(s)


this is caused by a bug in the 1.12.5 release (will be fixed in 1.12.6).
This is fixed already in our development tree at:

 - http://sofia-sip.org/repos/sofia-sip

You can get the latest sources with the darcs tool...

 cmd> darcs get http://sofia-sip.org/repos/sofia-sip

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[Sofia-sip-devel] sofia-sip meet up at IETF68?

2007-03-08 Thread Kai Vehmanen
Hello all,

the 68th IETF meeting will be held in Prague a couple
weeks from now (18-23 March 2007):

http://www3.ietf.org/meetings/68-IETF.html

I'm going there myself, and would be happy to meet up with any 
people using/interested in Sofia-SIP who are also attending (or 
otherwise in Prague at the time). Please drop me a mail and we can 
agree some free time slot.

PS And of couse, if it turns out there are many of us going,
   we could arrange a bigger common meeting as well.

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Re: [Sofia-sip-devel] sofia-sip-1.12.5 Compile failure on win32, VC 6

2007-03-06 Thread Kai Vehmanen
Hi,

On 06 March 2007,  [EMAIL PROTECTED] wrote:
>I tried it on 1.12.4 and 1.12.5 and got the same errors.  I 
>ran "sofia-sip\win32\autogen.cmd" before hand.  I did have to 
>edit build_sources.cmd because I was using gawk 3.1.3 (as the 
>comment in build_sources.cmd directs you to do).  I'll try 
>your latest code and report back.

aa, ok -- you could also try with mawk. Your error hints at a problem
with the autogenerated header files, so it could be an issue with
the version of gawk you are using. You can get mawk from:

http://gnuwin32.sourceforge.net/packages/mawk.htm

I've tested our current code with both gawk-3.1.5 and mawk-1.3.3,
and the code compiled fine.

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Re: [Sofia-sip-devel] sofia-sip-1.12.5 Compile failure on win32, VC 6

2007-03-06 Thread Kai Vehmanen
Hi,

On Mon, 5 Mar 2007, [EMAIL PROTECTED] wrote:

> I'm getting the following compiler errors (and more) when compiling 
> sofia-sip-1.12.5 win32, using Visual C 6:

hmmm, I just pushed a few VC6 specific fixes to our darcs tree, so it 
should compile again (will be in 1.12.6, but you can get the latest tree 
already now with "darcs get http://sofia-sip.org/repos/sofia-sip";). I'm 
not entirely sure whether 1.12.5 has these same issues (it probably does).

> C:\downloads\sofia-sip\sofia-sip-1.12.4.tar\sofia-sip-1.12.4\libsofia-sip-ua\msg\msg_basic.c(82)
>  : error C2065: 'msg_error_hash' : undeclared identifier
>
> C:\downloads\sofia-sip\sofia-sip-1.12.4.tar\sofia-sip-1.12.4\libsofia-sip-ua\msg\msg_basic.c(82)
>  : error C2099: initializer is not a constant
>
> C:\downloads\sofia-sip\sofia-sip-1.12.4.tar\sofia-sip-1.12.4\libsofia-sip-ua\msg\msg_basic.c(82)
>  : error C2065: 'msg_error_d' : undeclared identifier
>

Hmm, this is from 1.12.4, not 1.12.5...? :o Anyways, these look different 
from the errors I fixed today. Have you run 
"sofia-sip\win32\autogen.cmd" before building?

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Re: [Sofia-sip-devel] Probleme with machine state on INVITE request

2007-03-01 Thread Kai Vehmanen
Hi,

On Thu, 1 Mar 2007, moreauf wrote:

>   case nua_i_invite:
>   if (nh ==NULL) nh = nua_handle(nua,magic, TAG_END());
>nua_respond(nh,200,"OK", TAG_END());
>   break;;

I think the only thing you are missing is providing the SDP to use in the 
response. See for example:

- http://sofia-sip.org/repos/sofsip-cli/src/ssc_sip.c

... and search for "nua_respond"

>   case nua_i_bye:
>   if (nh ==NULL) nh = nua_handle(nua, magic, TAG_END());
>   nua_respond(nh, 200, "OK", TAG_END() );
>   nua_handle_destroy(nh);

You don't need to respond separately to nua_i_bye with nua_respond() -- 
the stack will do this for you.

See the examples of terminating calls at:

- http://sofia-sip.sourceforge.net/refdocs/nua/

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Re: [Sofia-sip-devel] Threading and connection reuse

2007-03-01 Thread Kai Vehmanen
Hi,

On Thu, 1 Mar 2007, Michael Manousos wrote:

> 1) Threading model
>
> As I understand the stack offers the option to operate with one thread for 
> every
> call-leg or with one thread per call-leg (operation mode selected with the
> su_root_threading(...) function).

hmm, not quite. The su_root_threading() mechanism controls whether the 
stack state machine is run in a separate thread, or in the same context as 
the nua.h public functions are handled. But the stack never creates per 
call leg threads.

> 2) Connection reuse
>
> The stack does not implement connection reuse as per
> draft-ietf-sip-connect-reuse-xx.

No, not yet (patches are welcome ;)).

We do have some support for reusing persistant connections, mainly 
intended for the first client->server hop (draft-ietf-sip-outbound type of 
usage).

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Re: [Sofia-sip-devel] How do you match message with response?

2007-03-01 Thread Kai Vehmanen
Hi,

On Thu, 1 Mar 2007, Cristian Bidea wrote:

> How can you match a message that you send nua_register() or
> nua_subscribe() with the response received in an event of type
> nua_i_notify or nua_r_subscribe?

the nua handles (nua_handle_t) are meant for this purpose. You create a 
handle, give it to nua_register/subscribe/foo(), and later get the 
matching nua handle as a callback parameter.

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Re: [Sofia-sip-devel] Static build sofsip_cli

2007-02-27 Thread Kai Vehmanen
Hello,

On Mon, 26 Feb 2007, [EMAIL PROTECTED] wrote:

> try to build sofsip-cli-0.1.12 static so that it run on everyone  linux 
> computers
> without building gstreamer .

now this might turn out to be difficult. I don't know if it's at all 
possible to use use gstreamer without any dynamic libraries. Basicly all 
gstreamer processing elements are plugins loaded dynamically (= 
on-demand). So even if you link the core gstreamer libraries statically to 
the sofsip_cli binary, the gstreamer plugins would still be dynamic.

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Re: [Sofia-sip-devel] sofsip_cli call error

2007-02-22 Thread Kai Vehmanen
Hi,

On Wed, 14 Feb 2007, [EMAIL PROTECTED] wrote:

> i have sofsip on Nokia770 installed,
> sofsip can be started but if I liked to call get I this error:
[...]
> sofsip_cli: ssc_media_fsgst.c:618: priv_cb_ready: Assertion `depayloader != 
> ((void *)0)' failed.

that means you are missing the gstreamer depayloader elements (check the 
README file of sofsip-cli and the requirements for 'fsgst' media backend).

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[Sofia-sip-devel] 1.12.5 release aftermath, towards 1.12.6

2007-02-22 Thread Kai Vehmanen
Hello all,

it seems at least a few serious bugs slipped into the final 1.12.5 release 
(despite the heavy last minute patch'n'hack work). So let's freeze all 
work on new features and/or major refactoring, and focus on fixing known 
issues of 1.12.5, and then get release 1.12.6 out with these fixes.

In case someone has noticed this, our webpage is still showing info 
for the old 1.12.4 release. :( Unfortunately we are not able to modify 
the website now as sourceforge.net is having trouble with shell access:

  http://sourceforge.net/docs/A04

As for packaging issues, I today pushed patches for 1.12.5 release to 
Debian's pkg-voip repository on alioth. But as 1.12.5 has known bugs, and 
Debian is in deep-freeze mode for the 4.0 release, I'll probably won't 
even try to push 1.12.5 just yet to unstable.

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[Sofia-sip-devel] ANN: sofia-sip 1.12.5 released!

2007-02-12 Thread Kai Vehmanen
Hello all,

the first Sofia-SIP release for year 2007 is now finally out! Release 
notes follow:

Release notes for Sofia-SIP version 1.12.5 (2007-02-12)
===

Sofia-SIP is an open-source SIP User-Agent library, compliant with
the IETF RFC3261 specification. It can be used as a building block
for SIP client software for uses such as VoIP, IM, and many other
real-time and person-to-person communication services. The primary
target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a
SIP stack developed at the Nokia Research Center. Sofia-SIP is
licensed under the LGPL.

Changes since last release
--

Support for request queuing has been added to the high-level user-agent 
API. Various portability improvements have been made related to Linux, 
Mac OS X, Windows and Solaris ports. SIP registrations have been 
modified to utilize persistent transport connections. The GObject 
interface to 'nua' (nua-glib) has been dropped from the package and is 
now distributed separately. Severe bugs in 'su', 'nua', 'nta', 'stun' and 
'su-glib' modules have been fixed.

API/ABI changes and versioning
--

New features in API are marked with Doxytag macro @NEW_1_12_5.
Experimental features are marked with Doxytag macro @EXP_1_12_5.

libsofia-sip-ua:
- Added nua tags NUTAG_APPL_METHOD() and NUTAG_NEWSUB()
- Added nua_substate_make(), nua_substate_name()
- Added nta_incoming_create_response() function
- Added tport tags TPTAG_TOS(), TPTAG_DUMP(), TPTAG_LOG()
- Added tport predicate function tport_is_connected()
- Added authentication functions auc_info() and auc_has_authorization(),
   added type msg_auth_info_t for Authentication-Info header
- Added msg_header_join_items() function
- Added sip_is_allowed() function and k_bitmap field to the
   sip_allow_t structure 
- Added sl_header_log implementation, updated its prototype
- Added experimental SIP headers Suppress-Notify-If-Match and
   Suppress-Body-If-Match and functions related to them,
   enabled with ./configure option --enable-experimental
- Added SIP header Refer-Sub and related functions
- Added  include file
- Added auc_info() function (sofia-sip/auth_client.h)
- Added alternative implementations to event reactor object (su_port_t,
   referenced by su_root_t) that can be changed at runtime using SU_PORT
   environment variable, for instance
   - Internal semantics of su_port_t reference counting have changed:
 now su_port_create() has one reference, and su_root_create_with_port()
 always consumes one reference
- Changed return type of bm_memmem() and bm_memcasemem() to non-const
- This release is ABI/API compatible with applications linked against
   any 1.12.x release. However, applications built against this release won't
   work against an older library. The ABI has been tested with the nua module
   unit test (test_nua) built against original 1.12.0 release.

libsofia-sip-ua-glib:
- The 'nua-glib' module has been removed from the library and moved
   to a separate package 'sofia-nua-glib'. The remaining library (su-glib)
   is now considered stable and will be API/ABI compatible with later
   releases in the 1.12.x series.
- ABI has been modified and applications built against 1.12.4 and earlier
   releases, need to be rebuilt.
- Added su_glib_prefer_gsource() which makes glib-based su_port_t
   implementation the default choice when su_root_create() is called

Contributors to this release


- Petteri Puolakka (patch to stun)
- Mikhail Zabaluev (patch to su-glib mainloop integration)
- Michael Jerris (patch to url parsing # in sip/sips userpart)
- Colin Whittaker (TPTAG_TOS())
- Roman Filonenko (TPTAG_LOG(), TPTAG_DUMP(),
   patch to query DNS-servers with IP-Helper on win32)
- Remi Denis-Courmont (patch to network change API)
- Martti Mela (poll() emulation with select(), IPv6 and OS X fixes)
- Kai Vehmanen (persistent registrations, release management)
- Pekka Pessi (all the rest)

See the AUTHORS file in the distribution package.

Notes on new features
-

- The su_root_t reactor uses different implementation (epoll, poll or select
   on Linux), depending on SU_PORT environment variable.

- nua now supports request queuing, for instance, an application can send
   overlapping BYE and NOTIFY requests. The stack engine takes care of
   sending the new request only after the previous one has been responded.

- RFC 4488 defines the Refer-Sub header. Its datatypes, related functions and
   methods declared in  include file. The Refer-Sub
   header structure can be accessed from sip_t structure with sip_refer_sub()
   method, e.g.,

   if (sip_refer_sub(sip) &&
   strcasecmp("false", sip_refer_sub(sip)->rs_value) == 0) {
  /* Do not create implicit subscription */
   }

Re: [Sofia-sip-devel] uClibc Cross Compile glib.h Error

2007-02-12 Thread Kai Vehmanen
Hi,

On Mon, 12 Feb 2007, Jerry Richards wrote:

> I am trying to cross-compile sofia-sip for a MIPS architecture using the
> uClibc compiler.  When I try to build it, I get the following errors:

you appear to have a broken glib installation. You can compile sofia-sip 
without glib, but in your case glib _was_ detected by configure as...

> /vobs/psp_linux/tools/mipsel-linux-uclibc/bin/mipsel-uclibc-gcc -DHAVE_CONF
> IG_H -I. -I. -I../.. -I../../libsofia-sip-ua/su/sofia-sip -I../../libsofia-s
> ip-ua/su -I../../libsofia-sip-ua/su -I../..//opt/glib/glib-2.12.0 -I../..//o

... has /opt/glib/glib-2.12.0 in the header list. Either try to fix the 
glib installation, or compile without glib support ('./configure 
--with-glib=no').

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Re: [Sofia-sip-devel] Assert failure

2007-02-12 Thread Kai Vehmanen
Hello,

summarizing from #sofia-sip:

On Fri, 9 Feb 2007, Michael Jerris wrote:

> Traced this one through, it is hitting assert(ss->ss_state !=
> nua_callstate_calling); in nua_invite_server_report.  This seems to make
> no sense as how could something on the UAS side be in the calling state?

This issue seems to be rare, and nobody has been able to reproduce it 
again. Pekka had some ideas on what could be causing this, but no patches 
yet.

A good question is whether the assert() should be turned into a warning 
(good for bug hunting, but bad for real-life use), or kept as is.

Anyways, we'll skip this issue for 1.12.5 release (in order to get it out) 
and try to find a fix for 1.12.6. If anyone can create a test case 
triggering this, or reproduce steps, that would help a lot.

> #0  0x003490c2e21d in raise () from /lib64/tls/libc.so.6
> #1  0x003490c2fa1e in abort () from /lib64/tls/libc.so.6
> #2  0x003490c27ae1 in __assert_fail () from /lib64/tls/libc.so.6
> #3  0x002a96f5a20d in nua_invite_server_report (sr=0x0, tags=0x0) at
> nua_session.c:2046
> #4  0x002a96f43f05 in nua_server_report (sr=0x404b4900) at
> nua_stack.c:1525
> #5  0x002a96f4324e in nua_stack_process_request (nh=0x9ebcd0,
> leg=0x2a9852b440, irq=0x2a984a0470, sip=0x2a98625478) at
> nua_stack.c:1256

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Re: [Sofia-sip-devel] 1.12.5-release: feature-freeze

2007-02-09 Thread Kai Vehmanen
Hi,

unfortunately I ran out of time with the release today, but we are almost 
there. I tagged the tree already once, but then we found out that the OS X 
event loop takes 100% of CPU time, and that had to be fixed (as is fixed 
in current darcs tree).

Note that there are now multiple rel-sofia-sip-1_12_5 tags in the tree. If 
you fetch sources with this tag, you'll get the correct one (latest). If 
for some reason you want to access an older 1.12.5 tag, you'll need to use 
the --to-match with a hash to get it.

And then a few comments still (to myself):

On Wed, 7 Feb 2007, Kai Vehmanen wrote:

> Specific items:
>
> * the NOTIFY response memory leak reported by Colin

We were unable to reproduce this in time.

> * any build issues (one reported by Mike today related to linker
>   errors)

Solved.

> The aim is still to release 1.12.5 this week. Deadline for patches is
> Friday 10:00GMT. :)

We almost made it. :) Anyways, 1.12.5 final is now in the darcs tree, and 
I've even written the release notes, but you'll have to wait until Monday 
for the official tarball and website update.

A big thanks to everybody who helped to get the release togeter! And a 
good weekend everyone!

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Re: [Sofia-sip-devel] Top of darcs build issue.

2007-02-09 Thread Kai Vehmanen
Hello,

On Thu, 8 Feb 2007, Colin Whittaker wrote:

> Here's a patch to fix it.
> $ darcs whatsnew
> {
> hunk ./libsofia-sip-ua/su/su_epoll_port.c 39
> -#if HAVE_EPOLL
> -
> hunk ./libsofia-sip-ua/su/su_epoll_port.c 53
> +#if HAVE_EPOLL
> +
> }

thanks, this was fixed in the darcs tree.

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[Sofia-sip-devel] 1.12.5-release: feature-freeze

2007-02-07 Thread Kai Vehmanen
Hello all,

it's time to freeze the features for 1.12.5. Let's focus solely on 
bugfixing and solving other release critical issues.

Specific items:

* ABI check: I did this today and high-level unit tests
   from 1.12.0 are compatible with current darcs tree.

* the NOTIFY response memory leak reported by Colin

* any build issues (one reported by Mike today related to linker
   errors)

The aim is still to release 1.12.5 this week. Deadline for patches is 
Friday 10:00GMT. :)

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Re: [Sofia-sip-devel] build time options to disable stun/nth modules

2007-02-06 Thread Kai Vehmanen
Btw, if you are short on storage space, "-Os" helps quite a bit...

On Tue, 6 Feb 2007, Kai Vehmanen wrote:

> --disable-stun
> --disable-nth
>
> Together, these cut down the stripped libsofia-sip-ua.so by 160K.

By rebuilding with "-Os" (gcc-4.1.2), you'll save additional 264K, 
resulting in libsofia-sip-ua.so with size of 792K.

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[Sofia-sip-devel] build time options to disable stun/nth modules

2007-02-06 Thread Kai Vehmanen
Hello,

I now added a few build time options that allow to shrink the library 
size:

--disable-stun
Disables the STUN module (the public API), example clients, and
support for STUN keepalives in SIP signaling.

--disable-nth
Disables http and nth (transaction layer) modules. As collateral
damage, you lose support for SIP signaling over HTTP-CONNECT.

Together, these cut down the stripped libsofia-sip-ua.so by 160K.

Now let's see if people find these useful, before going further. The 
downside of adding these options is that the makefiles get more and more 
complicated, which again means more testing and maintenance work.

If continued, the next candidate would probably be "nea" (SIP presence 
support), or alternatively "nua" itself (for apps that prefer to use "nta" 
directly). OTOH, specialized applications can always customize sofia-sip 
builds themselves, so we don't necessarily have to support all possible 
variations in the official releases.

Note also that these options modify the library API/ABI, so you can't 
freely change back and forth if you for instance maintaining sofia-sip 
packages for some Linux distribution.

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[Sofia-sip-devel] call for testing (was: preparing for 1.12.5 release)

2007-02-05 Thread Kai Vehmanen
Hello list,

I've now put a snapshot tarball of the upcoming release available at:

http://sofia-sip.org/~vehmanek/sofia-sip-1.12.4work-20070502.tar.gz

Please test it on your development environment, with your apps, test 
cases, etc, and report any errors (especially regressions w.r.t previous
releases). If all is well, final 1.12.5 will be released later this week.


On Mon, 29 Jan 2007, Kai Vehmanen wrote:

> Hello all,
>
> it's again time to start preparing for a stable release. I'll be handling the 
> release manager duties this time. The plan would be to get 1.12.5 out on week 
> 6, i.e. next week.
>
> In other words, please test the current development version in darcs
> and report any regressions and other problems you notice.
>
> The major changes since 1.12.4 are:
>  - nua-glib has been dropped from libsofia-sip-ua-glib, and this
>also means the library ABI was changed (-> libsofia-sip-ua-glib.so.3)
>   -> note: main library ABI is still backwards compatible
>  - nua has been refactored to get rid of static stack timers,
>and to allow multiple concurrent requests
>  - severe bug with timer accuracy when running sofia-sip under
>glib mainloop (via libsofia-sip-ua-glib)
>  - various bug fixes
>
> Current draft release notes at:
>
> - http://sofia-sip.org/repos/sofia-sip/RELEASE
>

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Re: [Sofia-sip-devel] setting the IP TOS for SIP messages and RTP packets.

2007-02-05 Thread Kai Vehmanen
Hi,

On Mon, 5 Feb 2007, [EMAIL PROTECTED] wrote:

> now I try to compileren gst-plugins-good on my nokia 770,however I get this 
> error :
[...]
> Nokia770-39:/home/user/gst-plugins-good-0.10.5# ./configure
[...]
> checking for gcc... no
> checking for cc... no

there's no compiler nor libraries on the device, so you'll have to 
cross-compile the components on your development PC. I recommend using the 
Maemo SDK, which makes cross-compilation easy:

  o http://maemo.org/

For questions specific to Maemo SDK, and 770/800, please consult the 
maemo-developer mailing list, and their website/wiki -- there's plenty of 
information available!

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[Sofia-sip-devel] preparing for 1.12.5 release

2007-01-29 Thread Kai Vehmanen
Hello all,

it's again time to start preparing for a stable release. I'll be handling 
the release manager duties this time. The plan would be to get 1.12.5 out 
on week 6, i.e. next week.

In other words, please test the current development version in darcs
and report any regressions and other problems you notice.

The major changes since 1.12.4 are:
   - nua-glib has been dropped from libsofia-sip-ua-glib, and this
 also means the library ABI was changed (-> libsofia-sip-ua-glib.so.3)
-> note: main library ABI is still backwards compatible
   - nua has been refactored to get rid of static stack timers,
 and to allow multiple concurrent requests
   - severe bug with timer accuracy when running sofia-sip under
 glib mainloop (via libsofia-sip-ua-glib)
   - various bug fixes

Current draft release notes at:

  - http://sofia-sip.org/repos/sofia-sip/RELEASE

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Re: [Sofia-sip-devel] setting the IP TOS for SIP messages and RTP packets.

2007-01-29 Thread Kai Vehmanen
Hi,

On Mon, 29 Jan 2007, [EMAIL PROTECTED] wrote:

> I have installed sofsip_cli-0-12 and sofia_sip-1.12.1 on my linux System.
> I'm looking at implementing setting the IP TOS for SIP messages and RTP 
> packets.
[...]
> 2)for Rtp transfer, i dont have found wo sent the RTP-Packet.
>
> ./sofiasip sip:[EMAIL PROTECTED] --media-impl=gstreamer

all the RTP stuff is handled by gstreamer, so you have to modify the code 
in gstreamer elements. Specifically, you need to modify gstudpsrc.c 
(receiving) and gstdynudpsink.c (sending) in gst-plugins-good package:

http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-plugin-udp.html

One thing to watch out for that "--media-impl=gstreamer" uses gstudpsink 
for sending, while "--media-impl=fsgst" uses gstdynudpsink.

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Re: [Sofia-sip-devel] XML parsing readily available?

2007-01-29 Thread Kai Vehmanen
Hi,

On Mon, 29 Jan 2007, Jan Van den bosch wrote:

> Does Sofia-SIP provide functionality to parse an incoming XML payload (say,
> an open/closed presence event) or do I have to use an other library to
> accomplish this?

there's no XML parser in Sofia, so you'll have to use an external library. 
There are plenty of alternatives to choose from: libxml is one, provides a 
C API, and has similar licensing terms as sofia-sip...

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Re: [Sofia-sip-devel] memory size of sofia-sip?

2007-01-26 Thread Kai Vehmanen
Hi,

On  25 January, 2007 Pekka Pessi wrote:

>I guess we should split the stack in smaller pieces and use dl 
>to load some components, like resolver, TLS and STUN if needed:
>
>
https://sourceforge.net/tracker/index.php?func=detail&aid=1403134&group_id=1
43636&atid=756076

hmm, I'm still leaning more towards having build time options 
to (de)select features. For example:

 - if you don't need STUN keepalives, nor STUN-based discovery 
   of contact IP (in most cases rport/RFC3581 will do), 
   STUN can be safely disabled
 - SSL/TLS: if you don't need a) TLS as a SIP transport, and 
   b) STUN shared-secret support, and c) something else I've forgotton? :o
 - http stack: if you don't do XCAP, etc

... in desktop use, having these features in the library should not
be a problem to anyone (we are talking at some tens of Kbs of code),
and as a bonus we keep to maintain our current ABI. In embedded and
other code-size constraint uses, configuring the library with additional
build-time options is a not problem.

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Re: [Sofia-sip-devel] sresolv: (win32) using IP Helper to get DNSlist before querying the registry

2007-01-24 Thread Kai Vehmanen
Hello,

I now committed a modified version of the patch to darcs, which
has the IP helper support, but is disabled by default: you need
to modify win32/config.h, add iphlpapi.lib and compile.
Could someone verify this works as it should?

On 23 Jan 2007, Michael Jerris wrote:
>From: http://msdn2.microsoft.com/en-us/library/aa366073.aspx
>
>The IP Helper API is supported on Microsoft(r) Windows(r) 98, 
>Microsoft Windows Millennium Edition, on Microsoft Windows 
>NT(r) version 4.0 with Service Pack 4, Microsoft Windows 2000, 
>Microsoft Windows XP and Microsoft Windows Server 2003 family. 

thanks for looking those up. So it's pretty old stuff. Or in 
other words, my devel environment in ancient.

>Header Declared in Iphlpapi.h.
> 
>Library Use Iphlpapi.lib.
> 
>DLL Requires Iphlpapi.dll.  
>
>
>This api has been around at least as far back as IE5 (and I 
>believe in IE4), do you have at least the dll on your system?

Yups, I have the dll, but not the header file. I guess yet another 
reason to update my devel environment (vanilla VC6 plus IPv6 SDK)
on win32. Too bad the last VC6-supported platform SDK (feb 2003) is only
available via CD-purchase nowadays (which I'm too lazy to do). I'm now
trying to 
find someone who has the CD... 

A quick poll, are there any people here that would object deprecating
support for VC6? I mean iphlpapi.h works fine with VC6 if you have
the correct platform SDK, but I'm pretty sure I'm the only one left
using VC6 and if I upgrade, it's not very likely the builds will continue
to work with the compiler in the future...

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Re: [Sofia-sip-devel] sresolv: (win32) using IP Helper to get DNS list before querying the registry

2007-01-23 Thread Kai Vehmanen
Hi Roman,

thanks for the patch (and sorry for the late reply)!

On 16 Jan 2007, Roman Filonenko wrote:
>We have seen Windows to unexpectedly drop DNS related keys 
>(used by sresolv module) from the registry, while settings are 
>still available via IP Helper API. This leads to empty DNS 
>server list in the sresolv and non-functional application as a result.
[...]
>Seems that you already had some build problems with IP Helper 
>(for now I've added respective #include under HAVE_WINSOCK2_H)?

I've been trying out the patch today, and it's otherwise ok, but
I'm still wondering about the availability of IP Helper API. I 
found...

http://msdn.microsoft.com/library/default.asp?url=/library/en-us/iphlp/iphlp
/about_ip_helper.asp 

... but it doesn't mention which SDK's have it. My ancient
VC6 environment at least didn't. 

Anyways, I guess this could be an optional feature (we already
have code using IP-Helper in su/su_localinfo.c) -- but I'd need
feedback on selecting the default for win32/config.h. 

And I guess the code also needs to be made conditionally built (so 
that I can compile sofia-sip with my VC6 ;)).

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Re: [Sofia-sip-devel] sofsip_cli on nokia 770 - problems with voice communication

2007-01-17 Thread Kai Vehmanen
Hi,

On Wed, 17 Jan 2007, lukas wrote:

> I run this application in the fallowing way: ">sofsip-cli 
> --media-impl=fsgst" as the nokia770 doesn't support gstreamer media.
> Sending text messages works well between two devices, no problems with 
> p2p communication, but when i try to start voice communication the 
> fallowing problems occur:
[...]
> sofsip> tport_udp_error: Connection refused (111) [icmp type=3 code=3]
> reported by [192.168.2.3]:0 nta: INVITE (78316250): Connection refused (111)

seems like the INVITE is sent to the wrong address. Please rerun the tests 
with transport debugging enabled ("export TPORT_LOG=1 ./sofsip_cli") and 
mail me the logs and I'll have a look.

For more info about debugging tools'n'tricks is available at:
   - http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ
 see QA3

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Re: [Sofia-sip-devel] placing a call. Thread question

2007-01-05 Thread Kai Vehmanen
Hi Peter,

On Thu, 4 Jan 2007, Peter P wrote:

> I have problems with threads, main loop and so on (the simple
> SofiaTutorial works fine).
>
> It seems that to call I need two threads: one who will initiate the
> call and other  to receive the callbacks (I understood it from
> refdocs/nua, "Place a call" section -but it isn't a complete example
> :-( )

you don't necessarily need two threads, but you do need to run the Sofia 
mainloop in some thread. Sofia stack iself does not rely on threads 
(although it can optionally use them), but instead the whole stack is 
event driven (see the Reactor design pattern).

> I see that su_root_create is "blocking", so I need to use Glib
> functions to have two threads: who will initiate the call and who will
> receive the callbacks.

The su_root_create() function is not blocking, but su_root_run() is (see 
the code at SofiaTutorial wiki page).

The glib bindings are an alternative way to run the sofia stack (using the 
glib eventloop instead of Sofia's own event loop), but on Windows you most 
likely don't want to use glib at all. Instead you should run Sofia's own 
event root in some win32 thread.

You can either integrate your whole application to the Sofia event loop 
(register events, timers, etc to the Sofia root, and then run it), or 
alternavile have a another thread for GUI logic for example (this is 
probably needed on win32 as we don't have glue code to integrate Sofia to 
the win32 event loop).

On Mac OSX, Sofia also offers glue code to directly plug Sofia into the 
native event loop. On OS X, you can of course also use glib, or use the 
Sofia's own event loop. In the end it depends on the application 
requirements you have.

> After looking in lot of places (for example sofsip-cli) I notice that
> the main is to use su_root_source_create, sofsip_mainloop_create,
> sofsip_mainloop_run, etc.

Sofsip-cli uses glib's mainloop, but shouldn't follow this on win32.
Instread directly run the native Sofia event loop with su_root_run().

> I tried to "call" and then to receive the callbacks (executing
> su_root_run just after create the call) but has not worked. Is it
> possible to call without use threads?

You need to be running su_root_run() in order to make calls work. In a 
simplied test app, you can add a one-shot timer which will make the call. 
Then when you run su_root_run(), the stack is started and after a while 
your "make a call" timer callback is triggered.

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Re: [Sofia-sip-devel] hello!

2006-12-12 Thread Kai Vehmanen
Hi,

On Tue, 12 Dec 2006, Caesar Caligula wrote:

>  I am a student learning SIP. I have a question, can sofia-sip support the 
> windows os?

yes. Sofia-SIP can be used on Linux, BSD, Mac OS X, and yes, also on 
various Windows platforms. See the following links for more win32 specific 
info:

 * 
http://www.mail-archive.com/sofia-sip-devel@lists.sourceforge.net/msg01000.html
 * http://sofia-sip.org/repos/sofia-sip/win32/README.txt
 * http://sofia-sip.org/repos/sofia-sip/docs/devel_platform_notes.txt

PS Added this info also to
http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ

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Re: [Sofia-sip-devel] nta or nua

2006-12-12 Thread Kai Vehmanen
Hi,

On Mon, 11 Dec 2006, Yuling Duff wrote:

> I started look into sofia-sip but not sure between nua and nta. If our
> application only need a sip stack to process a few sip messages like
> invite, cancel, ack, options and bye, is nta module is sufficient? If so
> how to use it. Has anyone used just nta for their applications? Thanks
> for your help.

hopefully the following link answers to your question (Q1):

- http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ

There are AFAIK various apps using NTA (although NUA is more commonly 
used), but no major ones are released as open-source --- at least yet. If 
anyone on the list has a project using NTA, with sources available, do let 
us know so we can add your project as an example to the FAQ-Q1 above. :) 
The sofia-sip package does come with a few examples of how to use NTA.

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Re: [Sofia-sip-devel] anyone ever see windows clients lie about their bitrate?

2006-12-12 Thread Kai Vehmanen
On Fri, 8 Dec 2006, Mike Frantzen wrote:

> I realize this isn't strictly Sofia-SIP but it affects the RTP side of
> things built on Sofia.
>
> I've tried two windows clients, SJPhone and X-Lite and both lie about their
> bitrate.  Both are advertising a PCMU/8000 codec.  I watched the RTP packets
> in a sniffer and ran the numbers: SJPhone is actually sending 8400 bits per
> second (52.5 packets per second) and X-Lite is sending 7800 bits per
> second.  Both are running on the same windows box with a cheap embedded dell
> sound board.  The RTP stream with 8400 bits per second is causing a nasty 5%

I'm afraid this will always be the case. The sampling rates are _nominal_ 
sampling rates, and the real rates depend on the clock signals driving the 
ADC/DAC circuits. In all practical cases, the clock signals will differ at 
least slightly between the sender and receiver, so you will always have to 
prepare for clock drift. Additionally, many VoIP clients utilize other 
clock sources as well (like the PC system clock) for some things (let's 
say a separate thread that sends packets). As these clocks also have some 
error with respect to their reported nominal rates, the app might 
introduce even more drift to the system ... if the app developer doesn't 
know what he's doing. ;)

Of course, a good client should be able to generate a rate of packets 
close to the nominal rate, and ideally with minimal change over time. But 
on general purpose platforms (where the app developer cannot define what 
audio hardware is used), this is difficult to achieve. In many cases the 
errors can be significant and RTP receivers just have to be prepared to 
handle all sorts of timing errors in the incoming RTP stream.

> audio latency when discontinuous transmission is turned off; the latency
> grows at a rate of 3 seconds per minute since the windows client is sending
> 20ms worth of audio every 19.05ms.  The extra buffering is eventually fatal
> to the audio stream on my N770 when the DSP runs out of buffers (I think).

That's (19.05/20) is _pretty_ bad. :( Some el-cheapo soundcards can only 
reliably produce a few well-known sampling rates (48k is nowadays common 
for desktop uses), and running the cards with any other rates (like 8k for 
VoIP) will produce suboptimal timing. Many/most USB soundcards will 
introduce similar problems as they are not driven by the ADC/DAC clock, 
but by the USB bus clock.

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Re: [Sofia-sip-devel] Problem with HOLD/RESUME

2006-12-12 Thread Kai Vehmanen
Hi,

On Mon, 11 Dec 2006, [EMAIL PROTECTED] wrote:

> I have a problem making Hold and Resume with my sip calls. I'm using
> sofia-sip darcs version.
[...]
> When I want to make a hold, I call nua_invite with NUTAG_HOLD(1) to send
> a re-invite with "sendonly" added  in the media attibute of the SDP
> parameters, at this stage all work well.
> But when I Resume the call with NUTAG_HOLD(0), the re-invite is sent but
> without media attribute added, normally it must add "sendrecv" to the
> media attributes no?

this is according to spec. An SDP media with no direction attribute is 
equal to one with an explicit "a=sendrecv" (RFC3264 sect 5.1).

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Re: [Sofia-sip-devel] branching off nua-glib

2006-12-05 Thread Kai Vehmanen
Hello all,

On Wed, 22 Nov 2006, Kai Vehmanen wrote:

> we've run out of time maintaining and polishing up nua-glib for the 2.0
> release, so I'm now proposing to branch it off to a separate package so
> that we can freeze the rest of the libsofia-sip-ua-glib for sofia-sip-2.0
> release.

as there has been no disagreeing voices, I've now moved forward with this 
plan. Following actions have been taken today:

  - the 'nua-glib' module has been removed from the sofia-sip
darcs tree
  - the same code has been released as a separate 'sofia-nua-glib'
module at sourceforge.net:
http://sourceforge.net/project/showfiles.php?group_id=143636
  - the libsofia-sip-ua-glib soname has been set to
libsofia-sip-ua-glib.so.3 (will effect binary builds as the
API/ABI has changed)
  - these changes will be part of the next 1.12.5 release

Now, the positive side to all this is that we are able to freeze the 
libsofia-sip-ua-glib API/ABI (libsofia-sip-ua is already frozen), and are 
thus getting very close to being able to release 2.0!

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Re: [Sofia-sip-devel] sofsip_cli : Error - "mulawenc" -> FAILED

2006-11-30 Thread Kai Vehmanen
Hello,

On Wed, 29 Nov 2006, maemo nokia wrote:

> We complied and installed sofiasip library and sofsip client successfully.
> When we try to run sofsip client on maemo, we get the following warning 
> message.
[...]
> sofsip_cli[7496]: GLIB MESSAGE default - This program is linked against 
> GStreamer 0.10.5
> sofsip_cli[7496]: GLIB DEBUG default - priv_verify_required_elements:191
> sofsip_cli[7496]: GLIB MESSAGE default - Verifying GST element "mulawenc" -> 
> FAILED
> sofsip_cli[7496]: GLIB MESSAGE default - Verifying GST element "mulawdec" -> 
> FAILED
> sofsip_cli[7496]: GLIB MESSAGE default - Verifying GST element "dynudpsink" 
> -> OK
> sofsip_cli[7496]: GLIB MESSAGE default - Verifying GST element "udpsrc" -> OK

the application was not enable to find the required gstreamer elements 
installed on the device. Did you run this on a OS2006 770, original 770 
distro, or in sbox (Maemo SDK)? The OS2006 version should have the needed 
'gst-plugins-good' package installed by default.

> sofsip_cli[7496]: GLIB MESSAGE default - Selecting media implementation: dummy
> sofsip> UA: unknown event 'nua_r_set_params' (23): 200 OK

As the required media elements were not found, sofsip-cli falls back to 
using a dummy media subsystem (only useful for testing purposes).

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Re: [Sofia-sip-devel] ipv6 support

2006-11-23 Thread Kai Vehmanen
Hello,

On Wed, 22 Nov 2006, Marc Blanchet wrote:

>  I've ported Asterisk to IPv6 and doing some IPv6 interop tests (to
> fix my implementation...). Using sofia-sip 1.12.3 and sofsip_cli
> 0.12. What I found so far:
> - can register and do invite over Ipv6 to asterisk, but media works
> only over IPv4.

yes, I'm afraid the media subsystem does not quite support IPv6. 
But shouldn't be a big task either. Basicly, you'd need to patch:

Basicly you'd need to patch:
  - sofsip-cli/src/ssc_media_fsgst.c:farsight_netsocket_bind_udp_port()
  - either
   - src/ssc_media_fsgst.c:priv_setup_rtpbin()
   - src/ssc_media_gst.c:priv_setup_rtpelements()
  - and pass the media address parameter from signaling (ssc_sip.c)
to the media object (the params are passed in
ssc_sip.c:priv_create_ssc_media())

Patches to fix these are very much welcome!

The signaling side should be ok with IPv6. We've also tested the stack 
with a proprietary media stack that support IPv6, but never did test with 
sofsip-cli.

> sofsip> r
> UA: REGISTER sip:[EMAIL PROTECTED]:b00:1:1::4] - registering address to 
> network
> sofsip> outbound(0x81e26e0): detected NAT: [3ffe:b00:1:2::98] !=
> 3ffe:b00:1:2::98
> UA: unknown event 'nua_i_outbound' (8): 101 NAT detected
>::tag_null: 0
>
> the strcmp in libsofia-sip-ua/nua/outbound.c  seems to be flawed in
> not removing the [] before comparing. special case for ipv6 addresses.

This should be fixed now in the devel tree. Thanks for the report!

> B) forcing sofia_sip to use ipv6 and sdp offer is v6, still
> sofsip_cli binds the RTP port on v4:
>
> % sofsip_cli --contact 'sip:[3ffe:b00:1:2::98]:5060' --media-addr
> '3ffe:b00:1:2::98' --proxy 'sip:[3ffe:b00:1:1::4]' 'sip:101@
> [3ffe:b00:1:1::4]'

Yep, this is because of the abovementioned shortcomings. The 'media-addr' 
is not currently passed to the media stack, and is only used in SDP 
offer/answer. :(

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Re: [Sofia-sip-devel] Sofia-sip-devel Digest, Vol 6, Issue 10

2006-11-22 Thread Kai Vehmanen
Hi,

On Fri, 17 Nov 2006, Vincent Zhou wrote:

> I am implementing a video conferencing software using sofia-sip library.
> I found there is some delay about 0.5s when do audio communication in
> LAN.
> Would anybody tell me if it's normal?
> Can sofia-sip realize the audio/video communication without delay?

0.5s is definitely a bit too long if within the same LAN.

As Sofia-SIP only takes care of the signaling, it doesn't have any control 
over the media latency. What RTP/media implementation are you using? The 
problem is probably there.

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[Sofia-sip-devel] branching off nua-glib

2006-11-22 Thread Kai Vehmanen
Hi,

we've run out of time maintaining and polishing up nua-glib for the 2.0 
release, so I'm now proposing to branch it off to a separate package so 
that we can freeze the rest of the libsofia-sip-ua-glib for sofia-sip-2.0 
release.

So, any users of nua-glib out there? The code will be of course still 
available, but you'll have to install multiple packages. No changes 
should be needed to your application sourcecode.

Comments and feedback is welcome!

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Re: [Sofia-sip-devel] sofia-sip changes (2006-11-21)

2006-11-22 Thread Kai Vehmanen
Hi,

On Wed, 22 Nov 2006, Chris Robson wrote:

> Anyone try developing a B2BUA.  And has anyone develop SOA examples yet?

we don't have any examples using the "NTA" layer, but Freeswitch is a good 
example of how to do this with "NUA". See http://www.freeswitch.org/ ...

PS And for a FAQ entry on NTA-vs-NUA, see
http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ

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Re: [Sofia-sip-devel] Any sample for win32 ?

2006-11-16 Thread Kai Vehmanen
Hi,

On Thu, 16 Nov 2006, Akin Ocal wrote:

> Have anyone tried to port sofia to win32 ?

sure, we've had a win32 port since start. Our official releases have
build files for VC6, you can build it under cygwin/mingw (although we 
are not testing each release with these), and we know people using newer 
VC versions with Sofia-SIP, so it should be all good.

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Re: [Sofia-sip-devel] sofia-sip changes (2006-10-20)

2006-10-23 Thread Kai Vehmanen
Hello,

On Sat, 21 Oct 2006, Pekka Pessi wrote:

>> hmm, this is a bit tricky. With the latest tree, I'm not getting a
>> nua_r_register after a succesful authentication anymore - should I?
> That is funny, I get nua_r_register or nua_r_unregister as usual. Here
> is some output from test_nua -A -B -C

uh oh, my bad -- the bug was in sofsip-cli (it destroyd the operation
handle at a wrong time).

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Re: [Sofia-sip-devel] sofia-sip changes (2006-10-20)

2006-10-20 Thread Kai Vehmanen
Hello,

On Fri, 20 Oct 2006, Sofia-SIP Darcs Changes wrote:

> Fri Oct 20 20:30:11 EEST 2006  [EMAIL PROTECTED]
>  * nua_stack.c: send nua_r_authenticate error events to the application.
>
>M ./libsofia-sip-ua/nua/nua_stack.c -6 +11

hmm, this is a bit tricky. With the latest tree, I'm not getting a 
nua_r_register after a succesful authentication anymore - should I? 
Returning 'nua_r_authenticate' is probably better, but many old apps rely 
on getting the nua_r_register to see whether auth succeeded or not.

How to fix this...?

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[Sofia-sip-devel] ANN: sofsip-cli-0.12 released!

2006-10-20 Thread Kai Vehmanen
Hello all,

the devel tree has gathered enough small changes, so I thought 
it's time to put another release out.

- tarball at:
   
https://sourceforge.net/project/showfiles.php?group_id=143636&package_id=179933
- signatures
   md5: c9b0c1dc325c32801775201be22e6207 sofsip-cli-0.12.tar.gz

Changes in release: 0.12 (2006-10-20)
-

- Added a build time options (see SOFSIP_USE_GLIB_EVENT_LOOP
   in src/sofsip_cli.c) to disable use of the glib eventloop,
   and use of sofia-sip's own event loop instead. Glib/object
   are still mandatory libraries, but this allows to better
   understand which code is dependent on glib/object.
- Moved streamer specific utilities from gsdp_utils.h to
   ssc_media_gst_utils.h, renamed gsdp_utils.h to sdp_utils.h
- Added summary output to the end of "./configure" runs.
- Added a welcome to interactive mode banner.
- Added 'set' (for settings) command to interactive mode.
- Fixed a bug which caused a segfault when CTRL-D (like with
   "echo h |./sofsip_cli") was received.
- Fixed a bug in rejecting an incoming call.
- Farsight is now disabled by default to avoid compilation
   problems with different Farsight versions. You can reenable
   it with "--with-farsight" configure option. This change
   does not affect the "fsgst" media implementation. See README
   for more details about the different implementations.

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Re: [Sofia-sip-devel] fix build without glib

2006-10-19 Thread Kai Vehmanen
Hi,

On Thu, 19 Oct 2006, Michael Jerris wrote:

> Patch attached

I took a bit different approach, but hopefully with the same results ;) -- 
builds without glib should now work (again). Thanks for reporting!

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Re: [Sofia-sip-devel] Stunc test client not working.

2006-10-10 Thread Kai Vehmanen
Hi,

On Tue, 10 Oct 2006, Kai Vehmanen wrote:

> ugh, it seems something and/or someone broke stunc in 1.12.3. :( I'll look

ok, found it and fixed. This was broken by Pekka's huge patchset:

Tue Sep 19 16:33:29 EEST 2006  [EMAIL PROTECTED]
   * Fixes for using size_t/ssize_t/su_socket_t and isize_t/usize_t/issize_t.

... so this only affects 1.12.3 (1.12.2 doesn't count). But basicly all 
STUN related functionality is broken in 1.12.3. :(

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Re: [Sofia-sip-devel] Stunc test client not working.

2006-10-10 Thread Kai Vehmanen
Hello,

On Mon, 9 Oct 2006, Dieter Tschanz wrote:

> ./stunc stunserver.org -b
>
> The output I got looks like
>
> assign_socket: local socket is bound to 0.0.0.0:33016
> main: stun_bind() failed

ugh, it seems something and/or someone broke stunc in 1.12.3. :( I'll look 
into this as soon as I have a free braincycle-slot to spare. Filed a bug 
item as well:

   
http://sourceforge.net/tracker/index.php?func=detail&aid=1574342&group_id=143636&atid=756076

> My question is how might I get the public IP with the stack? Is there any
> possibility to pass some stuninformation at nua_create time or later
> (without using a registrar)?

You can either learn it with STUN and/or from a SIP server (via the rport 
attribute). If you can figure out the public IP in some other way, in 
theory at least you can give it to the stack via the SIPTAG_CONTACT() 
param to nua_invite()/nua_register() [1]

[1] http://sofia-sip.sourceforge.net/refdocs/nua/nua_8h.html
 ... scroll down to "Populating SIP Request Message with Tagged
 Arguments"

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[Sofia-sip-devel] links: builds for ubuntu and gentoo

2006-10-09 Thread Kai Vehmanen
Hi,

distribution packages/builds are now available for both Ubuntu and Gentoo:

http://wiki.opensource.nokia.com/projects/SofiaBinaries

There used to be opensuse.org packages for sofia-sip as well, but those 
seem to have disappeared.

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[Sofia-sip-devel] libs returned by pkg-config

2006-10-09 Thread Kai Vehmanen

Hello all,

I today merged a patch from Rémi Denis-Courmont, which trims the list of 
libs returned by 'pkg-config --libs sofia-sip-ua'.


Now I'm still wondering whether this will break anything, and would 
welcome others to look at this. Anyways, the best rationale I found for 
this is:


  http://lists.freedesktop.org/archives/avahi/2005-November/000386.html

... which does make sense (and thus I applied the patch).

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Re: [Sofia-sip-devel] Windows, and *BSD build fixes.

2006-10-03 Thread Kai Vehmanen
On Tue, 3 Oct 2006, Michael Jerris wrote:

> Attached are several patches to fix build issues on multiple platforms.

Thanks, applied.

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[Sofia-sip-devel] sf.net trackers

2006-10-03 Thread Kai Vehmanen
Hello all,

I'd humbly ask everyone to take a quick look at our bug and feature-req
trackers at sf.net and:

http://sourceforge.net/tracker/?group_id=143636&atid=756076
http://sourceforge.net/tracker/?group_id=143636&atid=756079

- close items that have been fixed/handled
- update items that are no longer valid
- add items for bugs/features that have been discussed in the past
   but since then forgotten

In general, the idea is that not all bugs (nor feature-reqs) need to go 
through these trackers. But items that for some reason cannot be resolved 
immediately (within a few days), should be added to one of the trackers 
so that they are not forgotten.

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Re: [Sofia-sip-devel] Duplicate Contact: headers today...

2006-09-29 Thread Kai Vehmanen
Hi,

just a quick update: 1.12.2 was in fact released but Pekka's announcement 
mail never reached this list (the tarball is at sf.net though). Oh well, 
the release has so many bugs that we are going to silently bury it and 
continue quickly to 1.12.3. Pekka's going to send an update on the plan 
for 1.12.3...

Thanks for all who have been quick in reporting bugs to us!

  On Thu, 28 Sep 2006, Colin Whittaker wrote:

> With yesterdays darcs pull I now get duplicate Contact: headers.
> Anyone have any ideas ?
>

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Re: [Sofia-sip-devel] Broken nua_invite: error 900

2006-09-27 Thread Kai Vehmanen
Hi,

On Tue, 26 Sep 2006, Kaluakoi wrote:

> I have been trying for some time to get basic functionality working using
> Sofia; essentially send an INVITE to a Sip server (Asterisk) and have it
> "work". I continually get an error '900' which seems to be translated from
> an internal error '500'. As near as I can tell, that means I have no
> session. I really don't understand the meaning of this. I'm trying to create
> a session, therefore, I don't have one... yet...

btw, monitoring the raw messages sent/received is an excellent way to 
debug any SIP apps. How to do this with Sofia-SIP is common enough 
question, so I dedide to add it to our FAQ -- question number three at:

   http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ

But I guess in your case no messages would be seen. But this is also 
useful piece of information as it means the params you are passing to NUA
are somehow incorrect.

> It would be much appreciated If someone could please point out the error in
> this simple code.

Let's take a look:

> ss->nua_handle = nua_handle (nua, ss, SIPTAG_TO (to), TAG_END ());

This is sufficient to set the destination of the call.

> nua_invite (ss->nua_handle, NUTAG_URL (url),
> SOATAG_ADDRESS("127.0.0.1:8000"),
> NUTAG_EARLY_MEDIA (1), TAG_END ());

So here you no longer need to pass the NUTAG_URL(). For a full list
of options you can give to nua_invite(), see

   
http://sofia-sip.sourceforge.net/refdocs/nua/nua_8c.html#1427a8f538e34b4e1ab206722f9cba2f

The above was updated today btw.

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Re: [Sofia-sip-devel] Current darcs and Mac OS X

2006-09-27 Thread Kai Vehmanen
On Tue, 26 Sep 2006, Brian West wrote:

> Also it seems that su_os_nw.h is MIA.
>
> /usr/local/freeswitch/include/sofia-sip-1.12/sofia-sip/nua.h:42:32:
> error: sofia-sip/su_os_nw.h: No such file or directory

Hmm, this is a bug all right. The intention was not install su_os_nw.h as 
the API is not quite ready yet (to be part of the stable set). Anyways, 
su_os_nw.h is not needed in the public nua.h interface, so this can be 
fixed easily (pushed fix to darcs tree).

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Re: [Sofia-sip-devel] NUTAG_IDENTITY and contact header

2006-09-26 Thread Kai Vehmanen
On Fri, 22 Sep 2006, Michael Jerris wrote:

> Attached is the beginnings of a patch for this can someone point me
> in the right direction of what else I need to do.. I know I need to add
> the nua_respond_to_register still.. what I would like to do is just send
> the event up to the event loop and be able to make a decision if I need
> to send it as 407 or 200 I think that right now I am just autoresponding
> with 200.

So based on today's discussion:

  - NUATAG_ALLOW("REGISTER") to nua_set_params
  - nua_i_respond event
  - nua_respond_to_inbound_register() with params
- handle created with nua_i_register
- status (200 = ok, 30x = moved etc, 40x = auth req.)
- headers like Expires, Supported, etc (via tags)

... and from there onwards. Probably the above list is incomplete as 
well, as this is totally new scenario for the NUA code, but should be a 
start at least.

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[Sofia-sip-devel] notes from irc meeting 2006-09-26

2006-09-26 Thread Kai Vehmanen
Hello all,

for those who couldn't attend today's irc meeting, here are some notes:

- 1.12.2 will be released any moment now
 - pending issues: remaining compiler warnings and testsuite
   issues
- 1.12.3
 - main features will be the alternate-identity support for nua
   and inbound registrations over nua
 - released when above features stabilize (target next month,
   at latest in November ; Nov-Dec should be reserved for the 2.0
   release)
- inbound registrations
 - Mike J. will head the work and the rest of us will try to
   help as much as possible
 - basics: pass registrations to nua client, allow client to
   accept, challenge for authentication or deny (nua_register_response())
 - authentication and handling the registration bindings
   (expire timers etc) should be handled in the app
- nua identity
 - Pekka Pessi heading the work
 - drop the NUTAG_IDENTITY and instead make alternate-identity
   something you realize with special nua_handle's (the "virtual nua"
   idea by Anthony M.)
 - clients can make multiple registrations (different identity,
   different registrar/proxy/etc params) and each result in a separate
   nua_handle (which are all associated with one nua_t holding the
   actual sockets)
 - calls, messages, etc can be associated with a registration handle
   (and thus also an alternate identity)
 - inbound requests are mapped to the approapriate registration
   handle (=identity)
- more meetings!
 - we'll arrange a similar meeting within next two weeks to check
   how 1.12.3 is progressing and address possible problems

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Re: [Sofia-sip-devel] unable to see other users

2006-09-26 Thread Kai Vehmanen

Hello,

On Mon, 18 Sep 2006, nide nide wrote:


I'm using sofsip-cli 0.11 with libsofia 1.12.1 and I'm facing two different
issues:
1°) I have troubles with the publish command. When I register and
authenticate to server, I try the 'p' command but sofsip-cli answers "UA:
PUBLISH: 405 Method Not Allowed". Do I have to do something before
publishing? gaim's sip plugin seems to work with the same library.


my first guess would be that the PUBLISH is sent to the wrong address. Try 
setting TPORT_LOG=1 ("sh> TPORT_LOG=1 ./sofsip_cli", or use 
ethereal/wireshark) to get log of the SIP messages sent and check that the 
request line of the PUBLISH message looks ok.



2°) Second issue, I cannot find how to list users already registered on the
server. Is there a command available in sofsip-cli or can I create it
through the sofiasip library?


You mean list other users registered? Standard SIP servers do not provide 
a "list-all-registered" method, but you can subscribe to the status of 
people/addresses you know with the sofsip-cli 's sip:[EMAIL PROTECTED]' 
(as in subscribe) command.


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[Sofia-sip-devel] irc meeting 2006-09-26 17:00-18:00 EEST

2006-09-25 Thread Kai Vehmanen
Hello all,

to discuss some of the recent hot topics (support for multiple identity, 
registration handling in nua, release roadmap), let's have a irc meeting 
tomorrow, at 17:00 EEST (10am EDT, 4pm CEST) at #sofia-sip on 
freenode.org.

If this goes well, we could have more of these.

PS Hopefully the time is ok to everyone... I tried to pick a time based
on past activity at #sofia-sip.

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Re: [Sofia-sip-devel] [newbie] - NUA/NTA - FAQ

2006-09-20 Thread Kai Vehmanen

Hi,

On Wed, 20 Sep 2006, César Davi Ávila do Nascimento wrote:


I'm newbie on SOFIA-SIP, and I have a doubt about this affirmation:


welcome to the list!


" NUA has a limitation of supporting only one identity per one NUA instance"

What does it mean?
I'd like to implement a 8 channel SIP gateway using SOFIA-SIP. Could NUA
be used to do it?


A very good question as this has been the hot topic on this list for some 
weeks (if not longer). In short, currently the NUA API does not support 
this without hacks (otherwise you have to use NTA), but a lot of people 
are interested in fixing it. Some additional solutions (NUTAG_IDENTITY and 
NUTAG_M_*() tags) are under work (see archives for details):


- 
http://www.mail-archive.com/sofia-sip-devel@lists.sourceforge.net/msg00693.html
- 
http://www.mail-archive.com/sofia-sip-devel@lists.sourceforge.net/msg00759.html

... and others.

This really comes down to the design of your app. You could use the 
tapioca-sip approach of modifying "From" when 
requesting/answering, use the new NUTAG_M_USERNAME() tag to modify 
"Contact" per request/answer, run 8 nua instances (perfecly valid in some
cases), use the new NUTAG_IDENTITY() facility to handle multiple 
registrations with one nua instance, use NTA in place of NUA, or...


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[Sofia-sip-devel] comments on su_os_nw.h

2006-09-19 Thread Kai Vehmanen
Hello,

just took a quick glance at the su_os_nw.h API. I think at least the 
remove function should be modified before 1.12.2 release:

su_os_nw.h:
SOFIAPUBFUN int su_root_remove_network_changed(su_root_t *root);

... shouldn't the first arg be of type 'su_network_changed_t *' (to
be able to map create and remote requests)?

PS For Linux, probably the best way to implement this is
netlink NETLINK_ROUTE interface (see netlink(7) man page);
should be even doable without a separate notification thread.

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Re: [Sofia-sip-devel] NUTAG_IDENTITY and contact header

2006-09-19 Thread Kai Vehmanen
Hi,

forgot to reply this one.

On Thu, 14 Sep 2006, Michael Jerris wrote:

>> Of course, if someone has the time to try this out:
>>
>> - add a new "nua_i_register" event to NUA
>> - add "nua_respond_to_register()" public method
> This is exactly what we had been discussing doing, is there a good
> example of something we could base this off of, or are the 2 steps above
> all there is to it?

there are no examples available, and it's possible that there's more to it 
than just adding the above two new items to the API. So some 
investigation/design work is needed.

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Re: [Sofia-sip-devel] MinGW

2006-09-18 Thread Kai Vehmanen
Hello,

On Fri, 15 Sep 2006, Stanislav Zahariev wrote:

> When are you planning to implement full support for MinGW? The last time i

in principle we try to support both Cygwin and MinGW (and both have worked 
at least at some point), but we are not continuously testing on these 
platforms, so build errors are possible (and it's good to report these).

> tried using the latest gcc and CVS sofia-sip I get the following errors (btw
> I had to change -pthread to -pthreadVC2):

Hmm, should '-pthreadVC2' be always the default for MinGW?

> ./.libs/libstun.a(stun_dns.o): In function
> `priv_sres_cb':C:/msys/1.0/home/sofit/sofia-sip/libsofia-sip-ua/stun/stun_dns.c:88:
> undefined reference to `_imp__sres_sort_answers'
> :C:/msys/1.0/home/sofit/sofia-sip/libsofia-sip-ua/stun/stun_dns.c:121:
> undefined reference to `_imp__sres_resolver_timer'

This seems very strange. It looks like the linker cannot find functions of 
the 'sresolv' module... :o

What's the compiler command that produces the above error?

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Re: [Sofia-sip-devel] NUTAG_IDENTITY and contact header

2006-09-14 Thread Kai Vehmanen
On Thu, 14 Sep 2006, Michael Jerris wrote:

> It appears that from everything that is said, the current tags regarding
> user and identity tend to only work with something that we register to.

Yes, the case NUA has been designed for is:

  - one set of physical ports open
  - one public identity (with associated registrar and possibly proxis)
  - operations done with that identity

Of course our tag interface allows pretty broad interface for insert 
customing header values to messages, which can be utilized to cover more 
use-cases (as is done in tapioca-sip).

The NUTAG_IDENTITY would basicly expand NUA to cover:

  - not one, but multiple public identities
  - operations done with one of the maintained public identities

> What about something that we do not need or want to register too, but
> still need to serve up a specific identity and auth profile to when
> registering, ect.

Hmm, "do not want to register" and "when registering" in the same 
sentence...? :) Can you give a practical example? My guess what you want 
is NUTAG_IDENTITY() with optional registration (just a public identity).

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Re: [Sofia-sip-devel] NUTAG_IDENTITY and contact header

2006-09-14 Thread Kai Vehmanen
On Thu, 14 Sep 2006, Pekka Pessi wrote:

> I'd propose creating a separate proxy/registrar. They can run within
> same process, but proxy/registrar must use nta directly.

Of course, if someone has the time to try this out:

- add a new "nua_i_register" event to NUA
- add "nua_respond_to_register()" public method

This wouldn't be enough to implement a full-blown registrar, but might be 
sufficient for simpler uses like SIP clients with P2P add-ons. The NUA 
layer should not do any processing (maintain registration entries, etc), 
but just pass the messages.

Patches are welcome!

But, but, I second Pekka's comment, that with current APIs, building the 
code on top of NTA is the way to go.

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Re: [Sofia-sip-devel] NUTAG_IDENTITY and contact header

2006-09-14 Thread Kai Vehmanen
Hello again,

On Wed, 13 Sep 2006, Henrik Nordstrom wrote:

> Currently I use the same Contact for all identifies, and switching
> identity based on To. But I have found some registrars dislikes multiple
> identities registered with the same contact. I think being able to
> specify a contact like sip:[EMAIL PROTECTED]:port] in register would solve 
> some
> issues, and then reuse this in outgoing requests where needed.

as an alternative to Pekka's NUTAG_M_USERNAME(), you can also disable 
Sofia-SIP's outbound logic (that rewrites the contact) altogether 
(NUTAG_OUTBOUND("no-natify") to nua_set_params()), and then pass 
SIPTAG_CONTACT_STR() to all nua operations (nua_register(), nua_invite(), 
nua_respond(), etc).

To initialize the host:port part of the contact, you can query 
NTATAG_CONTACT_STR() in nua_r_get_params() callback. You need to call 
nua_get_params() after nua_create() to get this callback.

I'm not 100% certain this will work (the danger is that the stack will 
rewrite the Contact in some cornercase), so Pekka's NUTAG_M_USERNAME() 
might be more future-proof (and allows to use the NAT traversal logic).
But at least simple tests I did worked fine (I could pass my custom 
contact to nua_register() and that was also used in the message sent to 
network).

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Re: [Sofia-sip-devel] How to handle "unknown" nua events gracefully?

2006-09-14 Thread Kai Vehmanen
On a second thought...

On Thu, 14 Sep 2006, Kai Vehmanen wrote:

> So leaving the handles alone is more future-proof, but destroying them is 
> more robust. I guess a reasonable default is actually 
> nua_handle_destroy()'ing the handles of unknown events.

I came with a better idea: destroy the handle if it is a completely new 
one, otherwise ignore. This is future-proof, but does not cause handle 
buildup in nua. Documented also at our new FAQ page:

http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ

PS And implemented in the devel version of sofsip-cli (darcs repo).

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Re: [Sofia-sip-devel] How to handle "unknown" nua events gracefully?

2006-09-14 Thread Kai Vehmanen
Hi,

On Fri, 11 Aug 2006, Henrik Nordstrom wrote:

> What is the best method to handle unknown nua events gracefully?

that's a very good question. :)

> In the application callback I currently just log the event, and then
> call nua_handle_destroy() on the given handle.
[...]
> In the example cli app I see that the event is only logged. Doesn't this

This is tricky as an unknown event might be associated to an existing 
call/message and nua_handle_destroy()'ing the this case would terminate 
that as well. So for instance, a new nua event is added to report on call 
state in some future sofia-sip version...

> cause a possible buildup of "unused" nua handles? Or is that dealt with
> already by reference count making the nua_handle_destroy() call
> unneeded?

... but this could happen as well. The handles would not be leaked, but 
there would be buildup (until nua instance is destroyed).

So leaving the handles alone is more future-proof, but destroying them is 
more robust. I guess a reasonable default is actually 
nua_handle_destroy()'ing the handles of unknown events.

> And what about notifying the sender? Do you recommend that I respond
> with a 501 when seeing unknown messages?

This should be taken care by the stack. You enable/disable supported 
features at stack initialization time (tags like 
NUTAG_ENABLE_MESSAGE(bool) to nua_set_params()), and stack will respond 
with a 50x if these are received.

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Re: [Sofia-sip-devel] XCAP server

2006-09-13 Thread Kai Vehmanen
Hi,

On Wed, 13 Sep 2006, Henry Luan wrote:

> I want to use XCAP server to test my sip based endpoint. I'm wondering if
> there is any open source XCAP server available.

the only one I know of is the SER's XCAP module. Haven't ever used it, so 
cannot comment on how well it works:

http://ftp.iptel.org/pub/ser/presence/presence-handbook/

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Re: [Sofia-sip-devel] NUTAG_IDENTITY and contact header

2006-09-13 Thread Kai Vehmanen
Hello,

On Fri, 8 Sep 2006, Henrik Nordstrom wrote:

>> But for the former one, I guess you only need to add some custom tags to
>> the Contact header for each registration (multiple username parts maybe).
>> Brian, does this match your requirements? For this, something simpler than
>> NUTAG_IDENTITY() would be enough (or maybe already possible without
>> resorting to the 'nta' interface).
>
> Any further information on this? Especially how to do it in combination
> with NAT support..

none of this is implemented (although, some tricks might be possible even 
with current APIs), I'm just trying to understand what you need.

For making outbound calls, you might be able get away with adding a custom 
tag in the "Contact", and possibly other headers -> the recipient can 
identify which instance is calling. This should be possible by adding 
SIPTAG_FROM_STR(), SIPTAG_CONTACT_STR(), etc tags to nua_invite(), 
nua_respond(), etc. (like tapioca-sip does already).

For inbound calls, you either need multiple registrations (multiple public 
SIP identities registered), or no registrations at all (people will call 
to sip:[EMAIL PROTECTED], where 1.2.3.4 is your static IP, and "foobar" the 
instance identifier). For multiple registrations, you can do this by 
creating multiple nua_create() instances, or using the upcoming 
NUTAG_IDENTITY() feature.

For inbound registrations, you need to implement custom code on top of the 
Sofia-SIP NTA API.

Anyways, which of these scenarios you are interested in (and if all, which 
should be supported first ;))..?

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[Sofia-sip-devel] docs for libsofia-sip-ua-glib online

2006-09-13 Thread Kai Vehmanen
Hello,

I've finally fixed the doxygen rules for creating documentation for 
libsofia-sip-ua-glib. These are now also available online -- there are 
also some other updates to our developer links at:

  - http://sofia-sip.sourceforge.net/development.html

As always, comments are welcome.

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[Sofia-sip-devel] in or out-of-tree pthread-win32?

2006-09-11 Thread Kai Vehmanen
Hello all,

Michael raised a good point today at #sofia-sip: the current pthread win32 
DLL in our darcs doesn't really cut it, as it only works with one C 
runtime version.

Now any suggestions on how to best correct this? I guess we could just add 
a dependency that pthread-win32 [1] needs to be installed on the system 
when building Sofia-SIP (i.e. Sofia-SIP would not come with a copy of 
pthread). This way we wouldn't need to include the whole pthread-win32 
tree in our repository.

Pthread is only used in the 'su' (the portability wrapper of Sofia-SIP), 
so it's use is anyways pretty limited. nua_stack.h does include pthread.h, 
but I don't think it's really used (there's some conditionally included
code for refcounting).

I guess an alternative would be to implement the pthread-dependent parts 
of 'su' with native win32 threads. But I guess this wouldn't provide much 
benefits (at least at this point)... or what do you say?

[1] http://sourceware.org/pthreads-win32/

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[Sofia-sip-devel] sofia developer docs (was: Re: sofsip-cli updates, odd glib error)

2006-09-11 Thread Kai Vehmanen
Hello,

thanks Diego for the feedback! Some comments inline:

On Fri, 8 Sep 2006, Diego Betancor wrote:

> 1. Answer all questions. Few days ago I asked about a registrar (did read
> the doxygen generated documentation, read the mailing list) nobode
> answered. It didn't make me feel welcome.

Yes, I agree this is something very important. We should try to respond to 
all queries.

Btw, I guess in this case your mail was responded to in the end, but 
nevertheless, there have been other mails on this list that never got an 
answer, so we could do better.

> 2. Document the API in one place with examples. Not just put there the
> doxygen, because from it is not clear at all how to use. List main
> functions, list main tags in a point. Not only nua, but nta, su, soa . Lots
> of little examples. It can be in the wiki.

Yeps, this has been raised by others as well. Big thanks go to Nickolay 
for starting the tutorial page at our wiki.

I do hope that the growing supply of example/reference code will aid 
somewhat in the areas where our documentation is currently lacking. Also I 
think we should continue improving our doxygen documentation (especially 
to make it more easy to navigate). The doxygen docs have the big advantage 
that they are linked 1:1 to our version control system.

> 3. FAQ. Very important. How do I do this or that?. Why would I use nta and
> not nua?, what is the diference in taking care of dialog, ACKs, timers?.
> Could I use nua and nta at the same time?. How do I differenciate dialogs
> from diferent clients?, etc.

A good idea, let's start right away:
http://wiki.opensource.nokia.com/projects/Sofia-SIP_FAQ

> 5. How to help.

Another good one. First, I think we need help in getting our "how to help" 
page in shape. :) I put a start at:

http://wiki.opensource.nokia.com/projects/SofiaHowToHelp

I had more text already, but Firefox crashed (ugh, this was a first time 
for me with wikis -- I should know better), when I hit save. Feel free to 
improve and add things to the list.

> 6. Have all the documentation in a downloable tarball (or zip), so it can
> be read and studied locally.

Hmm, a good idea as well -- the doxygen stuff can be generated locally 
(and for instance on debian you can get it with 'apt-get install 
sofia-sip-doc'), we should probably advertize this option better.

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Re: [Sofia-sip-devel] compilation of Darcs version on Windows

2006-09-08 Thread Kai Vehmanen
Hi,

On Thu, 31 Aug 2006, Pekka Pessi wrote:

> On 8/31/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>> -
>> C:\Sofia_lib\sofia_sip_darcs\win32>gawk -v BINMODE=rw -f
>> ../libsofia-sip-ua/msg/msg_parser.awk success=-1 module=msg_test
>> NO_MIDDLE=1 NO_LAST
>> =1   PR=../libsofia-sip-ua/msg/test_protos.h
>> ../libsofia-sip-ua/msg/test_class.h  0> *** FAILED ***
>> -
>
> What is your awk version? (gawk --version says that mine is GNU Awk 3.1.3).

I now got the very same error with gawk 3.1.5. I fixed the problem by 
setting "success=0" instead of "success=-1" in build_sources.awk. Hmm, 
Pekka, why is the "=-1" there...? I traced the awk scripts and they seem 
to execute correctly...

-- 
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Re: [Sofia-sip-devel] sofsip-cli updates, odd glib error

2006-09-07 Thread Kai Vehmanen
Hello again,

On Fri, 25 Aug 2006, Henrik Nordstrom wrote:

> Just a small comment on sofsip-cli. imho it has grown a bit too large to
> be considered an example. Better considered an reference application
> using pretty much the complete stack.

ok, Nickolay also raised the same point. I've now updated the website a 
bit and sofsip-cli (and the gaim plugin) are now both classified as 
reference clients. There are now more prominent links to the tutorial as 
well as to the NUA reference documentation. Comments are welcome:

- http://sofia-sip.sourceforge.net/development.html

I've also update the front page somewhat:

- http://sofia-sip.sourceforge.net/

I added some new non-technical content to our 'Features' list. It'd be 
interesting to hear comments concerning the "Open development model" 
section. Is this project open enough?

-- 
  under work: Sofia-SIP at http://sofia-sip.sf.net

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