[pjsip] PJSIP based SIPek comaptibility with Realsip VOIP/SIP Server
hyd1_shikha
hyd1_shikha at sancharnet.in
Sun Nov 1 22:27:30 CST 2009
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
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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 at gmail.com>
Subject: [pjsip] 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) __________
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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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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 at gmail.com>
Subject: [pjsip] 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 at alice.it>
Subject: [pjsip] 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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