PJSIP based SIPek comaptibility with Realsip VOIP/SIP Server

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Dear Sir,


We are using  VOIP client application for Windows.We have used the Sipek application based on pjsip.
Sipek is a C# wrapper around PJSIP.

When we try to dial a call from Sipek client using the Realsip SIP Sever, the registration is successful, however, dialing fails.

We want to route our call through a VOIP provider via Internet.

Could you please suggest an existing server we can use which can be used instead of realsip which has been tested with pjsip and we can try the same.


Could you provide us guidance on an existing project we can use.

Could you also guide as to how we can encrypt the audio packet being sent using our own encryption alogorithm using pjsip and sipek.

Thanks in advance.

Regards,
Seema

  ----- Original Message ----- 
  From: pjsip-request@xxxxxxxxxxxxxxx 
  To: pjsip at lists.pjsip.org 
  Sent: Sunday, November 01, 2009 11:30 PM
  Subject: pjsip Digest, Vol 27, Issue 1


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  Today's Topics:

     1. Mod_pjsip (Meftah Tayeb)
     2. Freeswitch Pjsip Module (Meftah Tayeb)
     3. stun problem help request (giorge27 at alice.it)


  ----------------------------------------------------------------------

  Message: 1
  Date: Sat, 31 Oct 2009 20:27:53 +0000
  From: Meftah Tayeb <tayeb.meftah@xxxxxxxxx>
  Subject: Mod_pjsip
  To: "freeswitch-users at lists.freeswitch.org; pjsip list"
  <pjsip at lists.pjsip.org>
  Message-ID: <4AEC9DC9.30502 at gmail.com>
  Content-Type: text/plain; charset=ISO-8859-1; format=flowed

  Hi my FreeSWITCH friends
  Today we talked about FreeSWITCH libraries and we found pjsip, which is
  a lot better than sofia, it can do everything sofia can do and more,
  it's thread-safe and performs well, I would love to see a mod_pjsip.
  The library is GPL but they also offer alternative licensing, don't get
  me wrong, Sofia is great but I think FreeSWITCH will benefit from a
  thread-safe SIP library.
  don't say that i need some stuf for free, but we need to make fs a very
  also we love freeswitch to run in a Embedded PC like the blackfin
  CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open
  Source project
  With much love,
  DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus
  signature database 4539 (20091024) __________

  The message was checked by ESET NOD32 Antivirus.

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  http://www.eset.com




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  signature database 4539 (20091024) __________

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  http://www.eset.com




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  ------------------------------

  Message: 2
  Date: Sat, 31 Oct 2009 23:03:31 +0000
  From: Meftah Tayeb <tayeb.meftah@xxxxxxxxx>
  Subject: Freeswitch Pjsip Module
  To: pjsip at lists.pjsip.org
  Message-ID: <4AECC243.1050600 at gmail.com>
  Content-Type: text/plain; charset=ISO-8859-1; format=flowed

  hello,
  freeswitch is a nice Softswitch/PBX/Softphone Open Source Project.
  please could anyone contribute a Freeswitch Module that use Pjsip unstid 
  of Sofia SIP?
  sip only, no media.
  thank you!
  see you in Freenode Irc, #Freeswitch channel. my nickname is DelphiWorld.


  __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________

  The message was checked by ESET NOD32 Antivirus.

  http://www.eset.com





  ------------------------------

  Message: 3
  Date: Sun, 1 Nov 2009 15:47:09 +0100
  From: <giorge27@xxxxxxxx>
  Subject: stun problem help request
  To: <pjsip at lists.pjsip.org>
  Message-ID:
  <4AACEE901C38E1438250DDD2323BDEFD01A563 at FBCMST12V03.fbc.local>
  Content-Type: text/plain; charset="iso-8859-1"

  I don't understand the reason that d0nt' abilitate STUN server

  I use this code in python pjsip:
  userAgentCfg = pj.UAConfig()
  userAgentCfg.stun_host = "stun.voip.eutelia.it"
  lib.init(ua_cfg=userAgentCfg, log_cfg = pj.LogConfig(level=LOG_LEVEL,  callback=self.call_backLog))


  and i've always as result the stun server not abilitated and those logs:


  log_cb:  15:37:52.585   pjsua_core.c  STUN resolution failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT)
  log_cb:  15:37:52.597   pjsua_core.c  Ignoring STUN resolution failure (by setting)
  log_cb:  15:37:52.597   pjsua_core.c  pjsua version 1.4 for i686-pc-linux-gnu initialized
   
  i've tried even "pjsua" client with the option "--stun-srv=stun.voip.eutelia.it" and i've  the same behavior.
  In this case the log are:

   15:45:50.278   pjsua_core.c  Trying STUN server stun.voip.eutelia.it (1 of 1)..                    
  ^[[5~ 15:45:50.327       pa_dev.c  PortAudio sound library initialized, status=0                    
   15:45:50.327       pa_dev.c  PortAudio host api count=1                                            
   15:45:50.327       pa_dev.c  Sound device count=2                                                  
   15:45:50.328          pjlib  select() I/O Queue created (0x8e0a61c)                                
   15:45:58.181 stuntsx0x8e096  STUN timeout waiting for response                                     
   15:45:58.181    stunresolve  Session failed because STUN Binding request failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT)                                                                 
   15:45:58.181   pjsua_core.c  STUN resolution for stun.voip.eutelia.it failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT)                                                                    
   15:45:58.181   pjsua_core.c  STUN resolution failed: STUN transaction has timed out (PJNATH_ESTUNTIMEDOUT)                                                                                             
   15:45:58.199   pjsua_core.c  Ignoring STUN resolution failure (by setting)                         



  thank you
  Giorge


  Alice Messenger ;-) chatti anche con gli amici di Windows Live Messenger e tutti i telefonini TIM!
  Vai su http://maileservizi.alice.it/alice_messenger/index.html?pmk=footer
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