Re: [Sofia-sip-devel] 64-bit issue in sofia-sip (patch)

[Pekka Pessi]

2009/9/24 Aleksander Morgado :
> We just faced an issue happening only in 64-bit architecture, and found that
> it was already fixed by FreeSWITCH guys in their own sofia-sip repo:
> http://jira.freeswitch.org/browse/SFSIP-136
>
> It seems the fix is not included in the official repo, which would be good
> to have.

Thanks for noticing this. I've applied Stephan's patch.


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Re: [Sofia-sip-devel] problem shutting down nua

[Pekka Pessi]

2009/9/23 Francesco Lamonica :
> hi ppl, i am having some problems in shutting down the nua; i am still
> using sofia 1.12.9 and i am handling the eventloop with su_root_step.
>
>  I read on the list  the procedure pessi suggested:
>  1. call nua_shutdown(mynua)
>  2. wait for nua_r_shutdown with status code 200 then
>  3. call nua_destroy
>  4. break out of event loop (this should not be necessary in my case right?)
>  5. then call su_deinit()
>  my problem is that i never get the nua_r_shutdown, with any status

> what can i check?

That sounds weird. Best you can do is to enable more detailed
debugging output (e.g., env vars NUA_DEBUG=9 NTA_DEBUG=9) and try to
see where the execution hangs within the stack.

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Re: [Sofia-sip-devel] New release schedule?

[Pekka Pessi]

2009/9/23 Aleksander Morgado :
> Is there any planned date for a new stable (1.12.11) release of sofia-sip?

My current plan is to make a next release after we (Maemo devices
within Nokia) have stable Maemo 5 release. That should happen Real
Soon Now. However, there are few patches by Mikhail Zabaluev, who is
responsible for telepathy-sofiasip, that I'd like to include in the
stable release.

At the same time I'm planning to ditch Darcs as the principal VCS and
start using git. I'll have to check if it is possible to host
sofia-sip on maemo.org or should I set up something different.

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Re: [Sofia-sip-devel] Question aboiut NTA and sip_t

[Pekka Pessi]

2009/9/9 Daniel Corbe :
> When NTA receives a request which doesn't match an existing transaction it 
> passes a pointer to the callback function of type sip_t.  Can this pointer be 
> dereferenced through the lifetime of the transaction or does it go poof as 
> soon as I exit the callback function?

As Aleksander said, you can use sip_t as long as the transaction is alive.

The sip_t pointer lives inside an abstract msg_t container and you can
also get a new reference to it with nta_incoming_getrequest(). After
you are done with your sip_t / msg_t reference, you should remove the
reference with msg_destroy().

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Re: [Sofia-sip-devel] sresolv: DNS response code 5 (refused) not handled / Fallback to second DNS server not working

[Pekka Pessi]

2009/9/18 Timo Bruhn :
> [...] When sofia-sip tries to register its user agents at startup, the first 
> dns server is used
> to resolve the server name. The server refuses the request. Now the expected 
> behaviour
> should be a fallback to the second server. Unfortunately this never happens.

Yes, the fallback is probably the right thing to do.

> One way to fix this is to mark the server as temporarily unavailable as I did 
> in the patch
> attached to this mail. The fix is working fine in my test environment.

> Could anyone of the developers please have a look at it to verify that it 
> does not break
> anything else i did not see?

It should not break anything, or at least I don't think it does, but
here is my patch that also falls to the next server on server error
and
 not implemented error. Please give it a try, and report if it fixes
your problems (or crashes and burns), too.

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--- old-sofia-sip/libsofia-sip-ua/sresolv/sres.c	2009-09-24 13:16:38.0 +0300
+++ new-sofia-sip/libsofia-sip-ua/sresolv/sres.c	2009-09-24 13:16:38.0 +0300
@@ -3503,6 +3503,14 @@
 sres_send_dns_query(res, query);
 query->q_retry_count++;
   }
+  else if (error == SRES_AUTH_ERR ||
+	   error == SRES_UNIMPL_ERR ||
+	   error == SRES_SERVER_ERR) {
+dns->dns_icmp = res->res_now;
+sres_cache_free_answers(res->res_cache, reply);
+/* Resend query/report error to application */
+sres_resend_dns_query(res, query, 0);
+  }
   else if (!error && reply) {
 /* Remove the query from the pending list */
 sres_remove_query(res, query, 1);
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Re: [Sofia-sip-devel] Modify sofsip cli to work with PCMA instead of PCMA

[Pekka Pessi]

2009/9/14 fabien comte 
> I want sofsip cli works with PCMA instead of PCMU. How to modify it ?

Just in case you have not figured out it yet: you have to change the
RTP payload type on m= line to 8, too. The rtpmap should work,
provided the payload number on m=audio line and a=rtpmap are same, at
least in principle, but most implementations just look the payload
type number for well-known codecs like PCMU or PCMA.

> I modified ssc_media_gst.c [...] I tryed
> ->
>   self->sm_pt = 8; /* PT=0 => PCMA */
>   /* step: initialize the PT<->caps hash table */
>   pt_caps = gst_caps_new_simple ("application/x-rtp",
>  "clock-rate", G_TYPE_INT, 8000,
>  "encoding-name", G_TYPE_STRING, "PCMA",
>  NULL);
>
>   /* step: describe capabilities in SDP terms */
>   /* support only G711/PCMA */
>   caps_sdp_str = su_strcat(home, caps_sdp_str,
>   "v=0\r\n"
>   "m=audio 0 RTP/AVP 0\r\n"
>   "a=rtpmap:8 PCMA/8000\r\n");

   "m=audio 0 RTP/AVP 8\r\n"
   "a=rtpmap:8 PCMA/8000\r\n");

>   *dest = strdup(caps_sdp_str);
>   su_free(home, caps_sdp_str);
> It does not work (FAILED with 501 Not Implemented Yet)

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Re: [Sofia-sip-devel] How to use NUA iso NTA for basic registrar and calls

[Pekka Pessi]

2009/9/14 EiSl 1972 :
> I would like to use the NUA abstraction layer for my very simplified SIP
> call 'generator', which is of itself a kind of PBX (from SIP client point of
> view). Maybe call it an 'announcement server', calling each client when
> there is something to say :)
>
> Basically my requirements are:
> - Allow incoming registrations from multiple SIP clients
> - Being able to make calls (INVITE) to those registered clients, where I
> already know the media endpoints on my side. So I sent INVITE with valid SDP
> entries.
> - Being able to reINVITE to those clients
> - Being able to end calls (CANCEL / BYE)
> ==> it is no problem if the contact-info or from-field of the sent INVITEs
> equals for all clients.
>
> My point is:
> I would like to use NUA for this rather than NTA and then using the
> test_proxy.c as reference.
> Maybe I'm very wrong, but I'm under impression that this could be possible
> when I was checking the developers archive referring to introduction to
> nua_i_register and the discussion "NUTAG_IDENTITY and contact header"
> somewhere middle 2006.

Yes, it is possible to use nua as registrar. As far as i can recall,
you should include the tag NUTAG_ALLOW("REGISTER") with nua_create()
or some initial nua_set_params() call in order to get nua_i_register
events. As you said, there is some implementation on registrar
protocol in test_proxy.c. It is basically just the rules on accepting
and updating the Contact URIs and their expiration times.

> I'm aware that NUA is meant to be an UserAgent abstraction, but in principle
> it could still acts as a user-agent towards each registered client.
> If this is possible, can someone provide some guidelines? This since there
> is no reference on how to do this. Especially with respect to how to use
> nua_invite to build the correct messages to each client.

> The realization of
> the register seems straight-forward, except that a (re)register from the
> same client results into new handle so this gives me the impression that NUA
> is just doing a pass-through and does not do any administration on these.
> I'm willing to put the gathered knowledge in some Wiki / help to be added to
> the Sofia online docs.

You should save AoR (in To header) and Contact URIs from each
nua_i_register event from your clients. You then use AoR/To header
from nua_i_register as To header in INVITE, and Contact URI as request
URI (passed in as NUTAG_URL()).

The proxy case is bit different, it uses request URI in incoming
message to look for an AoR and the forward the request to the Contact
URIs.

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Re: [Sofia-sip-devel] Wrong answer from sofia-sip

[Pekka Pessi]

2009/8/26 Bernhard Suttner :
> attached the network trace. You have to decode the first paket as SIP in 
> Wireshark (destination ip).

Contact is used when the callee send caller a BYE request (or OPTIONS
or UPDATE or INFO etc...).

The RFC 3261 (section 18.2.1) specifies how to send the response:
unless you happen to have maddr parameter, the response is sent to the
source address, which is 10.251.0.1. If you can include a maddr
parameter in the incoming INVITE Sofia should send the response to the
address specified in maddr parameter. So you could have request like
this:

INVITE sip:757-2...@10.251.0.101:12004 SIP/2.0
Via: SIP/2.0/UDP 10.27.0.60:5060;branch=z9hG4bK3745033304-3162;maddr=10.27.0.60
From: "Aman DECT" ;tag=shorUA_3745033458-3162
To: 757-2026 
Contact: "Aman DECT" 
...

and Sofia would respond like this (to 10.27.0.60):

SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.27.0.60:5060;branch=z9hG4bK3745033304-3162;maddr=10.27.0.60;received=10.251.b0.1
From: "Aman DECT" ;tag=shorUA_3745033458-3162
To: 757-2026 
...



--Pekka

> -Ursprüngliche Nachricht-
> Von: Michael Jerris [mailto:m...@jerris.com]
> Gesendet: Mittwoch, 26. August 2009 16:53
> An: sofia-sip-devel
> Betreff: Re: [Sofia-sip-devel] Wrong answer from sofia-sip
>
> Can you paste the actual sip trace here.
>
> Mike
>
> On Aug 26, 2009, at 6:45 AM, Bernhard Suttner wrote:
>
>> Hi,
>>
>> I have the following problem (using sofia sip 1.12.9):
>>
>> A (non sofia-sip) sends INVITE to B (user agent with sofia-sip)
>> B send back a Trying
>> B send Ringing and then a 200 OK
>>
>> Most important data for the INVITE from A:
>> Src IP: 10.251.0.1
>> Dest IP: 10.251.0.101
>> Contact: 10.27.0.60
>>
>> The Trying generated from sofia-sip has the data:
>> Src IP: 10.251.0.101
>> Dest IP: 10.251.0.1
>> VIA: SIP/2.0/UDP 10.27.0.60;branch=ydfasfasfsf;received=10.251.0.1
>>
>> The problem is, that B use the IP 10.251.0.1 as destination and not
>> the
>> contact of the INVITE (that would be 10.27.0.60). Also in the RINGING
>> and 200 OK sofia-sip will use the Src-IP and not the Contact addr.
>>
>> Can I somehow configure sofia-sip in that way, that it will use the
>> CONTACT addr instead of the Src IP?
>>
>> Thanks in advance!
>>
>> Best regards,
>> Bernhard Suttner
>>
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>



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Re: [Sofia-sip-devel] Mac OS X 10.6 build problems

[Pekka Pessi]

Any news on this? Perhaps you have something in /usr/local/lib linking
to iconv? What is the "intl" library?

Try grep -l iconv /usr/lib/* /usr/local/lib/* ...

--Pekka


2009/9/1 Michael Jerris :
> I am chasing down this same issue on my box.  I should have fixes sometime
> this week and will push patches for them when I have them.  Note, I did have
> a successfull snow lep build a few weeks back before they did gold master on
> a different box, still trying to chase down what changed sense then.
> Mike
> On Aug 31, 2009, at 7:03 PM, Daniel Corbe wrote:
>
> gmake[3]: Leaving directory
> `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib/su-glib'
> gmake[3]: Entering directory
> `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib'
> /bin/sh ../libtool --tag=CC --mode=link gcc  -g -O2   -o
> libsofia-sip-ua-glib.la -rpath /usr/local/lib -version-info 3:0:0
>  su-glib/libsu-glib.la -L/usr/local/lib -lglib-2.0 -lintl   -lssl -lcrypto
> -lz   -lpthread
> mkdir .libs
> grep: /usr/lib/libiconv.la: No such file or directory
> sed: /usr/lib/libiconv.la: No such file or directory
> libtool: link: `/usr/lib/libiconv.la' is not a valid libtool archive
> gmake[3]: *** [libsofia-sip-ua-glib.la] Error 1
> gmake[3]: Leaving directory
> `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib'
> gmake[2]: *** [all-recursive] Error 1
> gmake[2]: Leaving directory
> `/usr/local/src/sofia-sip-1.12.10/libsofia-sip-ua-glib'
> gmake[1]: *** [all-recursive] Error 1
> gmake[1]: Leaving directory `/usr/local/src/sofia-sip-1.12.10'
> gmake: *** [all] Error 2
> sh-3.2# cd /usr/lib
> sh-3.2# ls *iconv*
> libiconv.2.4.0.dylib    libiconv.2.dylib        libiconv.dylib
>
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Re: [Sofia-sip-devel] 2nd SIP INVITE for Fax Call- 200 OK port set to 0

[Pekka Pessi]

2009/8/31 Kaustubh Madhav PATWARDHAN :
> I am developing a fax gateway application using the Sofia SIP.
>
> Following is the sequence of SIP messages happening in that order between
> Calling and Called Gateways
>
> Both the gateways are running Sofia SIP
>
> Calling Gateay    Called Gateway
>
> 1) INVITE (audio 9000 RTP..)     -> 2) 200 OK (audio 9000 RTP...)
>   |
>   |
>  \/
> 4) 200 OK (image 0 UDPTL t38) <- 3) INVITE (image 9000 UDPTL t38)
>
> When the Called Gateway sends the second INVITE for fax data, the Calling
> Gateway sends 200 OK, with image port set to 0. Then sending of fax data
> fails. I do not know, why the 200 OK should send image with port set to 0 ?
> Could you please provide your valuable insight and expert advice on this?

You should probably replace your user SDP with m=image line when
responding to 2nd INVITE?

If your user SDP does not have m=image line, then the explanation is
simple, m=image 0 means that the proposed fax transmission media in
2nd INVITE got rejected.

If you do have m=image line, then you have two options:

* use SOATAG_ORDERED_USER(1) and include only m=image line in user SDP
* use SOATAG_ORDERED_USER(0) (default) and include m=audio and m=image
lines in user SDP

Or then there might be a bug, so an example of the your session
including the SDPs sent in all 4 messages along with debugging output
with env variables NUA_DEBUG=9 SOA_DEBUG=9 would be very helpful...

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Re: [Sofia-sip-devel] 64-bit issue in sofia-sip (patch)

[Michael Jerris]

Seems I forgot to pass this upstream, sorry.  I'll go back through and  
make sure there are not any straglers.

Mike

On Sep 24, 2009, at 3:55 AM, Pekka Pessi  wrote:

> 2009/9/24 Aleksander Morgado :
>> We just faced an issue happening only in 64-bit architecture, and  
>> found that
>> it was already fixed by FreeSWITCH guys in their own sofia-sip repo:
>> http://jira.freeswitch.org/browse/SFSIP-136
>>
>> It seems the fix is not included in the official repo, which would  
>> be good
>> to have.
>
> Thanks for noticing this. I've applied Stephan's patch.
>
>
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Re: [Sofia-sip-devel] 64-bit issue in sofia-sip (patch)

[Aleksander Morgado]

>
> > We just faced an issue happening only in 64-bit architecture, and found
> that
> > it was already fixed by FreeSWITCH guys in their own sofia-sip repo:
> > http://jira.freeswitch.org/browse/SFSIP-136
> >
> > It seems the fix is not included in the official repo, which would be
> good
> > to have.
>
> Thanks for noticing this. I've applied Stephan's patch.
>
>
>
Grreat, thanks!
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[Sofia-sip-devel] [ sofia-sip-Patches-2865713 ] Deferrable timer fixes

[SourceForge.net]

Patches item #2865713, was opened at 2009-09-24 16:24
Message generated for change (Tracker Item Submitted) made by mzabaluev
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=756078&aid=2865713&group_id=143636

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Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Mikhail Zabaluev (mzabaluev)
Assigned to: Nobody/Anonymous (nobody)
Summary: Deferrable timer fixes

Initial Comment:
>From the Maemo 5 branch:
These patches fix a crash in deferrable timer processing for a GLib mainloop.
Also, the NUA stack timer and the transport timer are set as deferrable. I hope 
this does not break anything :)

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[Sofia-sip-devel] [ sofia-sip-Patches-2865716 ] Don't force the default CA file for TLS

[SourceForge.net]

Patches item #2865716, was opened at 2009-09-24 16:29
Message generated for change (Tracker Item Submitted) made by mzabaluev
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=756078&aid=2865716&group_id=143636

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Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Mikhail Zabaluev (mzabaluev)
Assigned to: Nobody/Anonymous (nobody)
Summary: Don't force the default CA file for TLS

Initial Comment:
This patch removes the code that initializes the default CA file name for 
OpenSSL, if it is not set by the application. This causes TLS transport to fail 
if there is no such file available to the process.
In Maemo 5 and other distributions, OpenSSL has the default CA path set up just 
fine,

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Re: [Sofia-sip-devel] Making sofia request type AAAA in DNS Query

[Pekka Pessi]

2009/8/26 Jerry Richards :
> I am cross-compiling sofia-sip and I'm wondering if anyone has gotten
> sofia-sip to request type  addresses in its DNS query?  Right now I only
> see it do a type A request and if that fails, it gives up.  I would like it
> to attempt type A and if that fails, attempt type .

I'm afraid you have to fix the source code. The sa_res_order field in
nta_t object is used to determine the order, and it is never
initialized (and it defaults to nta_res_ip6_ip4). A tag for
determining it value is missing.

You can add its initialization in nta_agent_create().

> I can see the RES_OPTIONS and SRES_OPTIONS in the documentation.  Are these
> compile-time environment variables?  Or are they runtime environment
> variables.  This makes a big difference when cross-compiling.

They are environment variables, I'm sorry if the documentation is
unclear. However, they do not change the nta resolving order.

-- 
Pekka.Pessi mail at nokia.com

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[Sofia-sip-devel] sofia-sip changes (2009-09-24)

[Sofia-SIP Darcs Changes]

This posting was generated automatically from darcs repo
.

Wed Aug 26 20:49:16 EEST 2009  Mikhail Zabaluev 
  * tport/tport_tls.c: Don't force the CA file name if not set
  Ignore-this: aa2fb2e168be7afecde9b75986218243
  
  This makes TLS work if OpenSSL already has a good default search path.

M ./libsofia-sip-ua/tport/tport_tls.c -1

Thu Sep 24 18:42:45 EEST 2009  Pekka Pessi 
  * nua: added tests for NUTAG_AUTO302(), NUTAG_AUTO305()
  Ignore-this: 8ac666f00a12fadfd0c747bd53289899

M ./libsofia-sip-ua/nua/check_nua.c +2
M ./libsofia-sip-ua/nua/check_nua.h +1
M ./libsofia-sip-ua/nua/check_simple.c +130

Thu Sep 24 18:46:17 EEST 2009  Pekka Pessi 
  * nua: added NUTAG_AUTO302(), NUTAG_AUTO305()
  Ignore-this: b8c0c194aff5b03534970cf376cf6163

M ./libsofia-sip-ua/nua/nua_client.c +5
M ./libsofia-sip-ua/nua/nua_params.c +14
M ./libsofia-sip-ua/nua/nua_params.h -1 +8
M ./libsofia-sip-ua/nua/nua_tag.c +73
M ./libsofia-sip-ua/nua/sofia-sip/nua_tag.h +10

Thu Sep 24 17:15:58 EEST 2009  Pekka Pessi 
  * run_test_sresolv: testing patch for SFSIP-152
  Ignore-this: 71cf89c3e7872a7f040d15dcfa3971be

M ./libsofia-sip-ua/sresolv/run_test_sresolv -1 +1

Thu Sep 24 17:13:30 EEST 2009  Lee Verberne 
  * sres.c: handle end-of-line comments (SFSIP-152)
  Ignore-this: 8c79d2fbd6c0d4113c341c71ab615de2

M ./libsofia-sip-ua/sresolv/sres.c -3 +3

Mon Sep 14 19:41:13 EEST 2009  Mikhail Zabaluev 
  * iptsec/auth_client.c: fix an endless loop goof
  Ignore-this: f2ad98456178d96b569cad178454e11

M ./libsofia-sip-ua/iptsec/auth_client.c -1 +1

Thu Sep 24 10:48:45 EEST 2009  Stephan Knoblich 
  * su_uniqueid.c: don't use incorrect sizeof
  Ignore-this: 8a7e47ec94744cd2f9aba3d86c25efa3

M ./libsofia-sip-ua/su/su_uniqueid.c -1 +1

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Re: [Sofia-sip-devel] sresolv: DNS response code 5 (refused) not handled / Fallback to second DNS serve r not working

[Timo Bruhn]

> 2009/9/18 Timo Bruhn :
> > [...] When sofia-sip tries to register its user agents at startup, the 
> > first dns server is used
> > to resolve the server name. The server refuses the request. Now the 
> > expected behaviour
> > should be a fallback to the second server. Unfortunately this never happens.
> 
> Yes, the fallback is probably the right thing to do.
> 
> > One way to fix this is to mark the server as temporarily unavailable as I 
> > did in the patch
> > attached to this mail. The fix is working fine in my test environment.
> 
> > Could anyone of the developers please have a look at it to verify that it 
> > does not break
> > anything else i did not see?
> 
> It should not break anything, or at least I don't think it does, but
> here is my patch that also falls to the next server on server error
> and
>  not implemented error. Please give it a try, and report if it fixes
> your problems (or crashes and burns), too.
> 
> -- 
> Pekka.Pessi mail at nokia.com
> 
> 
Thanks for your patch, Pekka. 
The fallback works and I did not see any problems during my tests.

Timo


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