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PJSUA: Disabling loose routing adds the Route header to the end of SIP message

Hi all,

With PJSIP version 1.10, the --no-force-lr option in PJSUA makes the "Route" header to be added to the end of the message just before the "Content-length" header.
With loose routing enabled (default mode), the Route header is added just after the Via header.

Page 25 of RFC 3261 says that the relative order of the header fields is not significant normally, although required to be on top in places where proxy is used.
This will only help the proxy in parsing the SIP messages faster.

Is there any problem with this behavior like some implementations of proxy *require* them to be on top?

Thanks in advance,
Sundar

Ashraf Jaddo | 8 Feb 16:28
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HELP about background mode

Hello,
 
 
I am using PJSIP 1.12.. I did not run any special config when I compiled the libraries.. just compile for ARM6 and ARM7 then merge.. I am trying to get background working so:
 
 
- I used TCP instead of UDP as my protocol.
 
- I checked PJ_ACTIVESOCK_TCP_IPHONE_OS_BG and PJ_IPHONE_OS_HAS_MULTITASKING_SUPPORT in OS_autu.h and I see both of the defined as 1. I see both of them defined in a “#if defined(PJ_DARWINOS) && PJ_DARWINOS!=0” so I did not change anything...
 
- I added Required Background mode flag into my test App plist file
 
When I launch my app from xcode by hitting the run button, then I press home on the device to put the app in background mode I see logs from my registration function every 60 seconds as expected. and when I call the device I hit the on_incomming_call event as expected....
 
But when I launch the app from the device itself just click on its icon the background does not work..
 
if I tried to call the device after 60 seconds, Astersik will say the device is not registered which mean the background socket did not respond to the Keep alive message between the server and the device..if I removed the plist entry about background mode It does not work in both cases (launch from xcode and launch from device), so I think I am on the right path...
 
 
To confirm that my settings are good i added log success to “static void activesock_create_iphone_os_stream(pj_activesock_t *asock)” when it set kCFStreamNetworkServiceTypeVoIP and I see that log when I start the app from xcode but again I know it is working from xcode but not as a stand alone..
 
 
Am I missing something else?? I have been researching this issue an I am not sure what is missing? Anyone who did get background mode to work ??
 
 
Thanks,,,,,
Fadi Chehimi | 8 Feb 11:29
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Which architecture is being used?

Hi

I have a question regarding architecture build targets for iPhone.

I use the normal way to build PJSIP for an iphone:

$ ./configure-iphone
$ make dep && make clean && make

But in the terminal I do not see any indicator about the architecture
being built against. When i looked inside configure-iphone scrip i
noticed that ARCH is commented out

#ARCH="-arch armv6"

With this I get hundreds of CPU_SUBTYPE_ARM_ALL warnings but not sure
which architecture causes them to appear.

I would be grateful if some sheds some light on this

I am using XCode 3.2.6 and SDK 4.3.1 and PJSip 1.10

Cheers

Florian | 8 Feb 01:25
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Force RTP through Asterisk server

Hi,

I have searched Google and the mailing list archive extensively, but
haven't found useful results.

We have the following setup:
- client A behind firewall
- client B behind other firewall
- Asterisk directly accessible

A and B both register with Asterisk.

Now when calling, A and B will try to send RTP packets directly to
each other. For various reasons, we cannot use that feature and we
want to always use the Asterisk server as an RTP relay. How do we need
to set up the account and/or the call for that?

Thanks,
Florian

Yunus Ozcelik | 7 Feb 18:10
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cross compile for openwrt os_core_... selection

Hi,

I am compiling pjsip 1.10 for OpenWRT. OpenWRT compiles pjsip as package and it has a package Makefile. As it
is suggested as pjsip website, I am looking the auto-configuration output and also compile logs and I see
that os_core_unix.c is compiled while I am expecting os_core_linux_kernel.c to be compiled.

Is anyone has comment on that? Especially regarding if it is correct configuration or I need to fix something?

Regards,

Yunus Ozcelik

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How to modify VIA header for a dialog to make use of public address mapped by STUN

Hi all,

I am using pjsip and pjnath APIs to build my user agent.
I would like to REGISTER with the Registrar using the mapped address obtained by STUN.
The STUN mapping is partly ok (having issues with port mapping though, described in my other post).
The question is how do I change the VIA header in a SIP dialog, once STUN response is obtained?

I am able to build the FROM, TO and the CONTACT fields with the account information and the public address obtained (shown below).
But the VIA header seems to be using my local transport address. How do I change it to the STUN mapped address?

message log:

REGISTER sip:iptel.org SIP/2.0
Via: SIP/2.0/UDP 192.168.0.23:5060;rport;branch=z9hG4bKPjiHYmNLr3J9bm5TyUGMJuhp3nPiJRk4MO
Max-Forwards: 70
From: <sip:sundar2 <at> iptel.org>;tag=i7pQoYqcuJnNBdIDWGI3TyR.hn3zMtTN
To: <sip:sundar2 <at> iptel.org>
Call-ID: OBdDB3SowCcfiNeUsXjdMuRBU2j.KGSI
CSeq: 1650 REGISTER
Contact: <sip:sundar2 <at> xx.xx.xx.xx:5060>
Expires: 3600
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE
Content-Length:  0
--

registration request procedure followed:

1. create registration client with pjsip_regc_create()
2. initialize client registration structure with pjsip_regc_init()
3. set the transport to be used with pjsip_regc_set_transport()
4, set authentication credentials with pjsip_regc_set_credentials()
5. set authentication preferences using pjsip_regc_set_prefs()
6. set refresh delay with pjsip_regc_set_delay_before_refresh()
7. create register request with pjsip_regc_register()
8. send the registration request with pjsip_regc_send()

Should I delve a bit deeper to play with the headers myself to get it done or is there something am missing?

Thanks in advance,
Sundar

Mihail Laba | 6 Feb 21:00
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bug in pjsua_callback::on_incoming_call

Hello

I tested two accounts on different asterisk servers.
On incoming call callback acc_id always = 0 regardless of account.
After call pjsua_call_get_info(call_id, &ci), ci.acc_id = 0 also.

Best regards, Mihail Laba

David Artman | 6 Feb 18:34
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Modifying registration connection timeout


Hello all,

Is anyone aware of where to modify the registration connection timeout?  I'm looking to lower it from the ~30 secs it is now.  Thanks,

David Artman
Miljenko Cvjetko | 6 Feb 14:50
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Visual Studio 2010 support

Hi

How far is support for Visual Studio 2010?

I have managed to
  • remove all Mobile platform deps and
  • got to linking where I have to solve known issue when migrating
    from VCBuild to MSBuild


Warning    41    warning MSB8012: TargetPath
(pjproject-2.0-beta\pjsip-apps\build\.\output\libpjproject--Win32-vc8-Debug\libpjproject-i386-Win32-vc8-Debug.lib)
does not match the Library's OutputFile property value
(pjproject-2.0-beta\lib\libpjproject-i386-Win32-vc8-Debug.lib).
This may cause your project to build incorrectly.
To correct this, please make sure that
    $(OutDir)=.\output\$(ProjectName)-$(TargetCPU)-$(PlatformName)-vc$(VSVer)-$(Configuration)\
    $(TargetName)=$(ProjectName)-$(TargetCPU)-$(Platform)-vc$(VSVer)-$(Configuration)
and
    $(TargetExt)=.lib
property values match the value specified in
    %(Lib.OutputFile)=..\..\lib\$(ProjectName)-$(TargetCPU)-$(Platform)-vc$(VSVer)-$(Configuration).lib
       
C:\Program Files\MSBuild\Microsoft.Cpp\v4.0\Microsoft.CppBuild.targets    1151    6    libpjproject



Is someone working on VS2010 and how far is he/she?

regards

mel


-- Miljenko Cvjetko dipl.ing. ET Direktor/CEO Projektant rješenja/Solution Architect Razvojni programer/Senior developer Voditelj projekta/Project Manager IX južna obala 13 Kajzerica Zagreb T: 385 1 7775555 M: 385 91 557 447 3 F: 385 1 7779556 e: mcvjetko <at> holisticware.net w: http://www.holisticware.net
Ke (Kevin) Yu | 4 Feb 06:59
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PJSUA play back delay

Hi All,


For some special reason, I need to delay the audio play back for 1 second for both ends. I tried both snd_play_latency or snd_rec_latency in pjsua_media_config (media_config. snd_play_latency  = 1000 or media_config. snd_rec_latency  = 1000). However, I didn't see the latency I expected at all. Both ends still played back the audio instantly. That was good for the purpose of voice calling. But it didn't satisfy my special requirement. Could anyone share any thoughts on how I can do it?

Thanks!
Gaurav iPhone | 3 Feb 12:00
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Re: Assertion failed: (strm), function interruptionListener, file ../src/pjmedia-audiodev/coreaudio_dev.c, line 770.

Dear Dileep,Vuk,


Actually I was having the crashing problem in pjsip 1.8 and getting a lots of complaints regarding that so I decided to update the pjsip 1.8 to 1.12 yesterday I was able to do that. Now my voip app working fine and not getting any problem on ios 5.01.

so you should run the latest pjsip stack on your app.


Thanks 

On Thu, Feb 2, 2012 at 6:04 PM, Vuk Kristjan <Kristjan.Vuk <at> innbox.net> wrote:
Thanks Gaurav,
� � �I was able to compile PJSIP1.8 for IOS5 and use PJSIP�functionality�, but i see only this assertion failure. Do you think �using PJSIP 1.12 will fix the issue?

Thanks
Dileep

On Wed, Feb 1, 2012 at 3:28 AM, Gaurav iPhone <gauraviphone05 <at> gmail.com<mailto:gauraviphone05 <at> gmail.com>> wrote:
Hello Dileep,


PJSIP 1.8 does not support ios 5.

you need to use pjsip 1.12.

On Wed, Feb 1, 2012 at 4:45 PM, dileep r <dileep0206 <at> gmail.com<mailto:dileep0206 <at> gmail.com>> wrote:
Hi All.
� �I have been getting this assertion failure in PJSIP�
� � ���Assertion failed: (strm), function interruptionListener, file ../src/pjmedia-audiodev/coreaudio_dev.c, line 770.
� � � � � � � � � � � � � � � � � , even though i'm not using the any PJSIP media files. I'm using PJSIP version 1.8 compiled for IOS 5


�Any one can help me on this?

Thanks in advance

Dileep

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