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	<title>Comments for pjsip blog</title>
	<atom:link href="http://blog.pjsip.org/comments/feed/" rel="self" type="application/rss+xml" />
	<link>http://blog.pjsip.org</link>
	<description>Tracking development of pjsip and embedded SIP stacks/SDKs</description>
	<lastBuildDate>Sat, 21 Aug 2010 10:23:31 +0000</lastBuildDate>
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	<item>
		<title>Comment on PJSIP version 1.6 is released by liuxueyan</title>
		<link>http://blog.pjsip.org/2010/05/11/pjsip-version-1-6-is-released/#comment-11187</link>
		<dc:creator>liuxueyan</dc:creator>
		<pubDate>Sat, 21 Aug 2010 10:23:31 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=266#comment-11187</guid>
		<description>hi,PJ
   I use the pjsip to develop a voip tools, When I test this tools in China,it&#039;s OK. But it can not connect to server,when test it in Japan. can you tell me where could come out problem? Network in Japan, where need to do a specail settings?
 
   Thank you very much.</description>
		<content:encoded><![CDATA[<p>hi,PJ<br />
   I use the pjsip to develop a voip tools, When I test this tools in China,it&#8217;s OK. But it can not connect to server,when test it in Japan. can you tell me where could come out problem? Network in Japan, where need to do a specail settings?</p>
<p>   Thank you very much.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on PJSIP version 1.6 is released by Shamun</title>
		<link>http://blog.pjsip.org/2010/05/11/pjsip-version-1-6-is-released/#comment-11179</link>
		<dc:creator>Shamun</dc:creator>
		<pubDate>Sun, 01 Aug 2010 18:37:51 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=266#comment-11179</guid>
		<description>I am using it now. Great.</description>
		<content:encoded><![CDATA[<p>I am using it now. Great.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on PJSIP version 1.6 is released by Kaustubh</title>
		<link>http://blog.pjsip.org/2010/05/11/pjsip-version-1-6-is-released/#comment-11178</link>
		<dc:creator>Kaustubh</dc:creator>
		<pubDate>Tue, 20 Jul 2010 09:56:33 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=266#comment-11178</guid>
		<description>Hi,
I used pjsip 1.6 to build a sip client for Mac OSx.
It has a problem on 10.5.x ie Leopard.
The incoming sound is not routed through the Headphones, it is routed through the Mac mini&#039;s speaker output.
But the issue is not replication on 10.6.x ie Snow Leopard.

If anyone does get the solution please let me know.

-Kaustubh</description>
		<content:encoded><![CDATA[<p>Hi,<br />
I used pjsip 1.6 to build a sip client for Mac OSx.<br />
It has a problem on 10.5.x ie Leopard.<br />
The incoming sound is not routed through the Headphones, it is routed through the Mac mini&#8217;s speaker output.<br />
But the issue is not replication on 10.6.x ie Snow Leopard.</p>
<p>If anyone does get the solution please let me know.</p>
<p>-Kaustubh</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on pjsip version 1.5.5 released: bug fixes only by ipad case</title>
		<link>http://blog.pjsip.org/2010/01/11/pjsip-version-1-5-5-released/#comment-11177</link>
		<dc:creator>ipad case</dc:creator>
		<pubDate>Tue, 06 Jul 2010 14:34:02 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=259#comment-11177</guid>
		<description>new version is out ! yay. thanks for the infos!</description>
		<content:encoded><![CDATA[<p>new version is out ! yay. thanks for the infos!</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Introducing pjnath &#8211; Open Source ICE, STUN, and TURN for NAT Traversal by Raghav</title>
		<link>http://blog.pjsip.org/2007/04/06/introducing-pjnath-open-source-ice-stun-and-turn/#comment-11174</link>
		<dc:creator>Raghav</dc:creator>
		<pubDate>Wed, 30 Jun 2010 19:31:05 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/2007/04/06/introducing-pjnath-open-source-ice-stun-and-turn/#comment-11174</guid>
		<description>Hi , I am trying to implement the the turn server rfc using java. my client is pjsua and pjnath-client.. ... The problem is that I am sending all the attributed,etc to the client but the client gives the response as &quot; Response Authentication failed&quot; , I think that the HMAC-SHA1 value I am calculating on the server is not the same what the client is getting. Since they dont match the response is dropped. Please help how do I calculate the message integrity .
Hoping an urgent response .


Thanks a lot</description>
		<content:encoded><![CDATA[<p>Hi , I am trying to implement the the turn server rfc using java. my client is pjsua and pjnath-client.. &#8230; The problem is that I am sending all the attributed,etc to the client but the client gives the response as &#8221; Response Authentication failed&#8221; , I think that the HMAC-SHA1 value I am calculating on the server is not the same what the client is getting. Since they dont match the response is dropped. Please help how do I calculate the message integrity .<br />
Hoping an urgent response .</p>
<p>Thanks a lot</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Native iPhone SIP Client Based on pjsip Available on App Store: Open Source and Not Tied to any Provider by Perry Ismangil</title>
		<link>http://blog.pjsip.org/2009/02/19/native-iphone-sip-client-based-on-pjsip-available-on-app-store-open-source-and-not-tied-to-any-provider/#comment-11161</link>
		<dc:creator>Perry Ismangil</dc:creator>
		<pubDate>Wed, 26 May 2010 19:57:59 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=234#comment-11161</guid>
		<description>Hi Santhana,

Firstly pjsip is licensed under GPL. So you can make any application you want, as long you have an open source application.

If you don&#039;t want to open source your application then we do provide other licensing options. Contact licensing@teluu.com for more information.</description>
		<content:encoded><![CDATA[<p>Hi Santhana,</p>
<p>Firstly pjsip is licensed under GPL. So you can make any application you want, as long you have an open source application.</p>
<p>If you don&#8217;t want to open source your application then we do provide other licensing options. Contact <a href="mailto:licensing@teluu.com">licensing@teluu.com</a> for more information.</p>
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	</item>
	<item>
		<title>Comment on Native iPhone SIP Client Based on pjsip Available on App Store: Open Source and Not Tied to any Provider by SANTHANA</title>
		<link>http://blog.pjsip.org/2009/02/19/native-iphone-sip-client-based-on-pjsip-available-on-app-store-open-source-and-not-tied-to-any-provider/#comment-11160</link>
		<dc:creator>SANTHANA</dc:creator>
		<pubDate>Wed, 26 May 2010 13:59:05 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=234#comment-11160</guid>
		<description>Helo All,

From this i got the pjsip has been accepted by App store and it is used in many iphone apps too..

but can i make of pjsip for any commercial iphone application? I mean do i need to get any special license to use this in commercial application or???</description>
		<content:encoded><![CDATA[<p>Helo All,</p>
<p>From this i got the pjsip has been accepted by App store and it is used in many iphone apps too..</p>
<p>but can i make of pjsip for any commercial iphone application? I mean do i need to get any special license to use this in commercial application or???</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Python SIP User Agent (Softphone) by mauro_fix</title>
		<link>http://blog.pjsip.org/2007/01/24/python-sip-user-agentsoftphone-wrapper/#comment-11159</link>
		<dc:creator>mauro_fix</dc:creator>
		<pubDate>Thu, 20 May 2010 20:47:24 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/2007/01/24/python-sip-user-agentsoftphone-wrapper/#comment-11159</guid>
		<description>when i compile the python_pjsua proyect it launch this error:

LINK : fatal error LNK1181: no se puede abrir el archivo de entrada ‘..\..\lib\libpjproject-i386-win32-vc8-release.lib’

i have no idea what’s wrong

thank for your attention..</description>
		<content:encoded><![CDATA[<p>when i compile the python_pjsua proyect it launch this error:</p>
<p>LINK : fatal error LNK1181: no se puede abrir el archivo de entrada ‘..\..\lib\libpjproject-i386-win32-vc8-release.lib’</p>
<p>i have no idea what’s wrong</p>
<p>thank for your attention..</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on PJSIP version 1.6 is released by mauro_fix</title>
		<link>http://blog.pjsip.org/2010/05/11/pjsip-version-1-6-is-released/#comment-11158</link>
		<dc:creator>mauro_fix</dc:creator>
		<pubDate>Thu, 20 May 2010 17:49:00 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=266#comment-11158</guid>
		<description>when i compile the python_pjsua proyect it launch this error:

LINK : fatal error LNK1181: no se puede abrir el archivo de entrada &#039;..\..\lib\libpjproject-i386-win32-vc8-release.lib&#039;

i have no idea what&#039;s wrong

thank for your attention..</description>
		<content:encoded><![CDATA[<p>when i compile the python_pjsua proyect it launch this error:</p>
<p>LINK : fatal error LNK1181: no se puede abrir el archivo de entrada &#8216;..\..\lib\libpjproject-i386-win32-vc8-release.lib&#8217;</p>
<p>i have no idea what&#8217;s wrong</p>
<p>thank for your attention..</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on Native iPhone SIP Client Based on pjsip Available on App Store: Open Source and Not Tied to any Provider by David</title>
		<link>http://blog.pjsip.org/2009/02/19/native-iphone-sip-client-based-on-pjsip-available-on-app-store-open-source-and-not-tied-to-any-provider/#comment-11156</link>
		<dc:creator>David</dc:creator>
		<pubDate>Tue, 18 May 2010 10:42:27 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/?p=234#comment-11156</guid>
		<description>Have you heard a lot about SIP phones and don’t know how to use it for your business? Well, it’s very simple. 

You can use SIP phones with two techniques. The first way is by a Softphone. Softphone is software that allows calling by VoIP and has all features of a traditional telephone. As the name suggests soft phone is a graphical representation of a regular telephone on your PC. All you need is set of your PC, broadband connection and an audio system attached to PC. There are many VoIP service providers that offer this service for free. Just download anyone that is most popular with many satisfied users and install it on your screen. These softwares come with easy to follow instructions and are fast on installation. Once you are there, invite your business partners to join the Softphone and start conferencing. You may also use SIP phones for &lt;a href=&quot;http://www.numberstore.com/&quot; rel=&quot;nofollow&quot;&gt;0800 Numbers&lt;/a&gt; or 0844 Numbers routing.

Alternative you may use a hardware that is designed exclusively for SIP telephony. Using this hardware eliminates the use of PC but requires a broadband connection. The hardware set is very similar to our traditional telephones. The only drawback with this system is that there is a cost involved in buying hardware. However it will give you a feel of regular telephone call.</description>
		<content:encoded><![CDATA[<p>Have you heard a lot about SIP phones and don’t know how to use it for your business? Well, it’s very simple. </p>
<p>You can use SIP phones with two techniques. The first way is by a Softphone. Softphone is software that allows calling by VoIP and has all features of a traditional telephone. As the name suggests soft phone is a graphical representation of a regular telephone on your PC. All you need is set of your PC, broadband connection and an audio system attached to PC. There are many VoIP service providers that offer this service for free. Just download anyone that is most popular with many satisfied users and install it on your screen. These softwares come with easy to follow instructions and are fast on installation. Once you are there, invite your business partners to join the Softphone and start conferencing. You may also use SIP phones for <a href="http://www.numberstore.com/" rel="nofollow">0800 Numbers</a> or 0844 Numbers routing.</p>
<p>Alternative you may use a hardware that is designed exclusively for SIP telephony. Using this hardware eliminates the use of PC but requires a broadband connection. The hardware set is very similar to our traditional telephones. The only drawback with this system is that there is a cost involved in buying hardware. However it will give you a feel of regular telephone call.</p>
]]></content:encoded>
	</item>
	<item>
		<title>Comment on How to Use Your Nintendo DS as a Phone and Make Free Calls by Free SvSIP-VOIP-WIFI Call @ Nintendo-DS Console &#171; AndroidBoss</title>
		<link>http://blog.pjsip.org/2007/09/20/how-to-use-your-nintendo-ds-as-a-phone-and-make-free-calls/#comment-11155</link>
		<dc:creator>Free SvSIP-VOIP-WIFI Call @ Nintendo-DS Console &#171; AndroidBoss</dc:creator>
		<pubDate>Tue, 11 May 2010 17:39:02 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/2007/09/20/how-to-use-your-nintendo-ds-as-a-phone-and-make-free-calls/#comment-11155</guid>
		<description>[...] office in Redmond), is this possible?  Yes this is possible, while Googling the net we found some useful tips which would be shared with all,   SvSIP / PJSIP: a piece of software which has the power to convert [...]</description>
		<content:encoded><![CDATA[<p>[...] office in Redmond), is this possible?  Yes this is possible, while Googling the net we found some useful tips which would be shared with all,   SvSIP / PJSIP: a piece of software which has the power to convert [...]</p>
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	</item>
	<item>
		<title>Comment on Python SIP User Agent (Softphone) by _gm</title>
		<link>http://blog.pjsip.org/2007/01/24/python-sip-user-agentsoftphone-wrapper/#comment-11151</link>
		<dc:creator>_gm</dc:creator>
		<pubDate>Thu, 08 Apr 2010 06:05:28 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/2007/01/24/python-sip-user-agentsoftphone-wrapper/#comment-11151</guid>
		<description>Hi, i want to know how can i define a custom conf slot for buffered RTP data, i.e. i get sound input and output in a buffered RTP stream instead of sound card device.</description>
		<content:encoded><![CDATA[<p>Hi, i want to know how can i define a custom conf slot for buffered RTP data, i.e. i get sound input and output in a buffered RTP stream instead of sound card device.</p>
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	<item>
		<title>Comment on Securing VoIP: SRTP Support in PJSIP by Benny Prijono</title>
		<link>http://blog.pjsip.org/2008/01/25/srtp-support-in-pjsip/#comment-11148</link>
		<dc:creator>Benny Prijono</dc:creator>
		<pubDate>Tue, 23 Mar 2010 01:49:47 +0000</pubDate>
		<guid isPermaLink="false">http://pjsip.wordpress.com/?p=75#comment-11148</guid>
		<description>As the report says, the problem is with the test suite, not with the library itself.</description>
		<content:encoded><![CDATA[<p>As the report says, the problem is with the test suite, not with the library itself.</p>
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		<title>Comment on Securing VoIP: SRTP Support in PJSIP by saperski</title>
		<link>http://blog.pjsip.org/2008/01/25/srtp-support-in-pjsip/#comment-11147</link>
		<dc:creator>saperski</dc:creator>
		<pubDate>Tue, 23 Mar 2010 00:05:07 +0000</pubDate>
		<guid isPermaLink="false">http://pjsip.wordpress.com/?p=75#comment-11147</guid>
		<description>Unfortunately, the srtp library exhibits some portability problems - i.e. for sparc64 or arm one is likely to run into data structure alignment problems.

See for example: http://delicious.com/tag/srtp+alignment

--Saper</description>
		<content:encoded><![CDATA[<p>Unfortunately, the srtp library exhibits some portability problems &#8211; i.e. for sparc64 or arm one is likely to run into data structure alignment problems.</p>
<p>See for example: <a href="http://delicious.com/tag/srtp+alignment" rel="nofollow">http://delicious.com/tag/srtp+alignment</a></p>
<p>&#8211;Saper</p>
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		<title>Comment on PJSIP on Symbian Phone Works by sergios</title>
		<link>http://blog.pjsip.org/2008/01/16/pjsip-on-symbian-phone-works/#comment-11144</link>
		<dc:creator>sergios</dc:creator>
		<pubDate>Tue, 02 Mar 2010 11:18:05 +0000</pubDate>
		<guid isPermaLink="false">http://blog.pjsip.org/2008/01/16/pjsip-on-symbian-phone-works/#comment-11144</guid>
		<description>Hi, I want to know why G722 codec is not implemented in pjsip symbian port? (building with Carbide 2.0 + GCCE + S60V5 working 100%) 

Thanks,</description>
		<content:encoded><![CDATA[<p>Hi, I want to know why G722 codec is not implemented in pjsip symbian port? (building with Carbide 2.0 + GCCE + S60V5 working 100%) </p>
<p>Thanks,</p>
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