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    <link>http://gmane.org</link>
  </image>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15579">
    <title>Re: Error pjsip</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15579</link>
    <description>&lt;pre&gt;Hello, turns out there is a quicker way to make the sound work on ubuntu
12.04 :
http://rrizun.blogspot.com/2010/12/pjsippjsua-wsound-on-ubuntu-1010.html

2012/5/22 Andrés Vargas &amp;lt;andnovar&amp;lt; at &amp;gt;gmail.com&amp;gt;



&lt;/pre&gt;</description>
    <dc:creator>:D :D  magdalena algawam</dc:creator>
    <dc:date>2012-05-22T16:24:00</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15578">
    <title>Error pjsip</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15578</link>
    <description>&lt;pre&gt;Hi,

I have installed pjsip 1.14.0 in a fresh ubuntu 12.04 installation. I have
had several problems.
The first one was pjsip didn't find any sound device. I installed
opencore-amr0.1.2 and I recompiled alsa 1.0.25 then it worked.

The next problem was that sometimes the call works and sometimes it shows
the next message and the call fail:

10:07:50.141 os_core_unix.c  pjlib 1.14.0 for POSIX initialized
 10:07:50.142 sip_endpoint.c  Creating endpoint instance...
 10:07:50.142          pjlib  select() I/O Queue created (0x8a2f9b8)
 10:07:50.142 sip_endpoint.c  Module "mod-msg-print" registered
 10:07:50.142 sip_transport.  Transport manager created.
 10:07:52.341   pjsua_core.c  pjsua version 1.14.0 for
Linux-3.2.0.23/i686/glibc-2.15 initialized

Listening on 200.126.23.174 port 35754

 10:07:52.344   tsx0x8a7b884  Failed to send Request msg
REGISTER/cseq=33114 (tdta0x8a7a818)! err=70018 (gethostbyname() has
returned error (PJ_ERESOLVE))
 10:07:52.344    pjsua_acc.c  SIP registration failed, status=502
(getho&lt;/pre&gt;</description>
    <dc:creator>Andrés Vargas</dc:creator>
    <dc:date>2012-05-22T15:19:55</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15577">
    <title>PJLIB thread/pools</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15577</link>
    <description>&lt;pre&gt;I am getting a 120012 from pj_thread_create  after about 380 threads, most
of them have been terminated when I get the error.

To create the thread:

pj_thread_t *thread=NULL;
pj_pool_t *threadPool=NULL;

pj_status_t SocketHandler::startThread(pj_pool_factory *pf,int index) {
    pj_status_t status;
    char name[1024];

    sprintf(name,"Client:%08x",index);

    // create pool
    threadPool=pj_pool_create(pf, name, PJ_THREAD_DEFAULT_STACK_SIZE,
PJ_THREAD_DEFAULT_STACK_SIZE, NULL);
    if (threadPool==NULL) {
        PJ_LOG(3,(__FILE__,"Unable to create memory pool"));
        return 1;
    }
    // create thread

status=pj_thread_create(threadPool,name,threadProc,this,PJ_THREAD_DEFAULT_S
TACK_SIZE,0,&amp;amp;thread);
    if (status!=0) {
        PJ_LOG(3,(__FILE__,"pj_thread_create returned %d",status)); //
this is where the error is logged
    }
    return status;
}


When destroying the threads:
  pj_status_t status;

   status=pj_thread_destroy(thread);
    if (status!=0) {
        PJ_LOG(3,(__FILE__,"pj_threa&lt;/pre&gt;</description>
    <dc:creator>Tom Johnson</dc:creator>
    <dc:date>2012-05-22T15:04:17</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15576">
    <title>Also PJSIP 1.14.2 released</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15576</link>
    <description>&lt;pre&gt;Hi everyone,

We've just also managed to make 1.14.2 release. Release note can be
found here: https://trac.pjsip.org/repos/milestone/release-1.14.2.

Again, thanks for all the contributions, feedbacks, etc!

BR,
nanang

&lt;/pre&gt;</description>
    <dc:creator>Nanang Izzuddin</dc:creator>
    <dc:date>2012-05-22T12:31:24</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15575">
    <title>Re: account.contact vs account.reg_contact in pjsua</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15575</link>
    <description>&lt;pre&gt;Hi,

I think you're not supposed to send the Contact with the sip.instance
stuff, see the example flow in
http://tools.ietf.org/html/rfc5626#section-9.5. Frankly I've forgotten the
exact reason. Just read the RFC to find out.

Best regards,
 Benny

On Fri, May 18, 2012 at 5:18 PM, Gustavo Garcia Bernardo &amp;lt;ggb&amp;lt; at &amp;gt;tid.es&amp;gt; wrote:

&lt;/pre&gt;</description>
    <dc:creator>Benny Prijono</dc:creator>
    <dc:date>2012-05-22T12:07:01</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15574">
    <title>Re: sample of custom transports</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15574</link>
    <description>&lt;pre&gt;Hi,

That has been a known issue: https://trac.pjsip.org/repos/ticket/1019, and
unfortunately there's no fix for it yet.

Best regards,
 Benny


On Tue, May 22, 2012 at 4:27 AM, Gary Metalle &amp;lt;Gary.Metalle&amp;lt; at &amp;gt;ipfx.com&amp;gt; wrote:

&lt;/pre&gt;</description>
    <dc:creator>Benny Prijono</dc:creator>
    <dc:date>2012-05-22T11:58:25</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15573">
    <title>Version 2.0 is officially released</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15573</link>
    <description>&lt;pre&gt;Dear all,

Finally it's done, after many months in the making, 2 alphas, a beta, and
an rc, pjsip version 2.0 is now released.

Please visit http://www.pjsip.org/ for more info.

Thanks everyone who's been giving us feedbacks, reports, questions,
patches, etc. Sorry if we're not able to respond to all of them, but now
that 2.0 work is done hopefully it'll be easier for us to find time here.

Best regards,
 Benny
&lt;/pre&gt;</description>
    <dc:creator>Benny Prijono</dc:creator>
    <dc:date>2012-05-22T11:36:03</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15572">
    <title>problems with pjsua</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15572</link>
    <description>&lt;pre&gt;Hello everyone,

I have just intsalled a newer version of ubuntu - 12.04 and i use pjsua
1.12, when i run the test wizard i get "no audio device" error message, any
idea how to fix that?

regards

Magdalena Algawam
&lt;/pre&gt;</description>
    <dc:creator>:D :D  magdalena algawam</dc:creator>
    <dc:date>2012-05-22T11:24:32</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15571">
    <title>To implement Proxy Server</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15571</link>
    <description>&lt;pre&gt;Dear All,

I am using Pjsip in our VOIP  application . Now I have to implement Proxy
Server in our registration process . So please guide me how to implement
Proxy Server.


Thanks &amp;amp; Regards
Chandra Bhushan Singh
9540322253
&lt;/pre&gt;</description>
    <dc:creator>chandra bhushan singh</dc:creator>
    <dc:date>2012-05-22T07:14:09</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15570">
    <title>sample of custom transports</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15570</link>
    <description>&lt;pre&gt;Hi.

Does anyone have an example of using custom transports? I've noticed that pjsip only allows a single TCP transport even if you try to create a new one for a different local/public address (i.e. a different network adapter). If you're using UDP  you can create as many different transports as you like.

I'd like to do this so that I can control which network adaptor is being used for a user account (of which there may be more than one).

I could live with having only one TCP transport at any one point in time if necessary, but I've had trouble with that. There isn't an API to make changes to a transport once you've created it and I've had problems trying to remove a transport so I can create a new one using:

pjsua_transport_set_enable(transport_id, PJ_FALSE);
pjsua_transport_close(transport_id, PJ_FALSE);

// bla

pjsua_transport_create(required_type, &amp;amp;new_cfg, &amp;amp;transport_id);

To be fair, the documentation does warn against calling pjsua_transport_close()anyway, but if you don't call it then you get pro&lt;/pre&gt;</description>
    <dc:creator>Gary Metalle</dc:creator>
    <dc:date>2012-05-21T21:27:28</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15569">
    <title>on_transport_state in transport tls</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15569</link>
    <description>&lt;pre&gt;When I use tls as transport, the transport state callback is not
initialized properly. I don´t receive any transport event, so I can´t
know when the transport is shutdown, for example.

I' m  using tls over openssl, and the method pjsip_tls_transport_start
in the file sip_transport_tls_ossl.c is executed without errors.

Any idea?.

Thanks!!!

&lt;/pre&gt;</description>
    <dc:creator>Jesús Gumiel</dc:creator>
    <dc:date>2012-05-21T20:47:03</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15568">
    <title>Re: help please</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15568</link>
    <description>&lt;pre&gt;Hello!
So, if you have libpjproject-i386-Win32-vc8-Debug.lib - you can run PJSUA
project (set it as start up project) and use PJSIP library under command
line. Can you do that?
&lt;/pre&gt;</description>
    <dc:creator>Fedot Fedotov</dc:creator>
    <dc:date>2012-05-21T15:40:46</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15567">
    <title>Trac error</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15567</link>
    <description>&lt;pre&gt;As of right now I'm getting the following error when trying to access http://trac.pjsip.org/repos/

Trac Error

TracError: The Trac Environment needs to be upgraded.

Run "trac-admin /home/autohost/pjsip/trac_wrapper/trac upgrade"


--
Saúl Ibarra Corretgé
AG Projects




&lt;/pre&gt;</description>
    <dc:creator>Saúl Ibarra Corretgé</dc:creator>
    <dc:date>2012-05-21T12:41:10</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15566">
    <title>help please</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15566</link>
    <description>&lt;pre&gt;Hi,
I have identify the error in following lines from different section of the
code.

pjsua_call_get_vid_stream_idx(call_id);
pjsua_vid_codec_set_priority(&amp;amp;pj_str("H264/97"),0);
pjsua_enum_aud_devs(aud_dev_info, &amp;amp;count);
pjsua_vid_enum_devs(vid_dev_info, &amp;amp;count);

This above lines are related to video support. I have added the library
libpjproject-i386-Win32-vc8-Debug.lib successfully in the
microsip project. But failed to compile. Please help..

BR,
Sazzad


On Fri, May 18, 2012 at 10:47 PM, sazzad hossain &amp;lt;sazzadjoy&amp;lt; at &amp;gt;gmail.com&amp;gt;wrote:

&lt;/pre&gt;</description>
    <dc:creator>sazzad hossain</dc:creator>
    <dc:date>2012-05-18T16:48:54</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15565">
    <title>Re: help please</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15565</link>
    <description>&lt;pre&gt;Hi,
I have identify the error in following lines from different section of the
code.

pjsua_call_get_vid_stream_idx(call_id);
pjsua_vid_codec_set_priority(&amp;amp;pj_str("H264/97"),0);
pjsua_enum_aud_devs(aud_dev_info, &amp;amp;count);
pjsua_vid_enum_devs(vid_dev_info, &amp;amp;count);

This above lines are related to video support. I have added the library
libpjproject-i386-Win32-vc8-Debug.lib successfully in the
microsip project. But failed to compile. Please help..

BR,
Sazzad

On Fri, May 18, 2012 at 2:02 PM, Fedot Fedotov &amp;lt;fedot.fedotov&amp;lt; at &amp;gt;gmail.com&amp;gt;wrote:

&lt;/pre&gt;</description>
    <dc:creator>sazzad hossain</dc:creator>
    <dc:date>2012-05-18T16:47:48</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15564">
    <title>Re: account.contact vs account.reg_contact in pjsua</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15564</link>
    <description>&lt;pre&gt;Reading at the documentation:
"reg_contact
Contact header for REGISTER. It may be different than acc contact if
outbound is used"

But why do they need to be different instead of always sending the
sip.instance/reg-id tags in all the Contact headers?

Regards,
G.


On 16/05/12 22:51, "Gustavo Garcia Bernardo" &amp;lt;ggb&amp;lt; at &amp;gt;tid.es&amp;gt; wrote:



Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo.
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_______________________________________________
Visit our blog: http://blog.pjsip.org

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pjsip&amp;lt; at &amp;gt;lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
&lt;/pre&gt;</description>
    <dc:creator>Gustavo Garcia Bernardo</dc:creator>
    <dc:date>2012-05-18T10:18:52</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15563">
    <title>Re: help please</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15563</link>
    <description>&lt;pre&gt;Hello!

You can:
1) build PJSIP library with build in PJSUA command line application, or
2) use another application in PJSIP source code
pjproject-2.0-RC\pjsip-apps\src\vidgui
it based at QT SDK for UI, but can help with understanding logic of PJSIP
using.
3) build PJSIP and integrate it into MFC-based dialog(as I do :-) )

in any case I think that build PJ libraries -
libpjproject-i386-Win32-vc8-Debug.lib(and release) - the quickest way to
prepare your own program with SIP client, based at PJlibrary.
&lt;/pre&gt;</description>
    <dc:creator>Fedot Fedotov</dc:creator>
    <dc:date>2012-05-18T08:02:37</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15562">
    <title>AS-SIP Support</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15562</link>
    <description>&lt;pre&gt;Hello,
I am currently looking into integrating a SIP stack with support for Assured Services on an embedded system using PPC and running Linux. I am able to compile PJSIP along with PJSUA for the PPC embedded system running linux and use PJSUA to make SIP communication from a host computer. However, PJSIP does not have the Assured Services support I was looking for. I was wandering if you have any suggestions on where I can start to implement the AS-SIP support? Are there any other open source SIP stacks I can use with AS-SIP support already implemented?
Regards,
Solan
&lt;/pre&gt;</description>
    <dc:creator>Solan Chala Bongase</dc:creator>
    <dc:date>2012-05-17T16:48:00</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15561">
    <title>Re: help please</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15561</link>
    <description>&lt;pre&gt;Shouldn't you be asking the MicroSip people this?

-----Original Message-----
From: pjsip-bounces&amp;lt; at &amp;gt;lists.pjsip.org [mailto:pjsip-bounces&amp;lt; at &amp;gt;lists.pjsip.org]
On Behalf Of sazzad hossain
Sent: Thursday, May 17, 2012 5:33 AM
To: pjsip&amp;lt; at &amp;gt;lists.pjsip.org
Subject: [pjsip] help please

HI,
I have tried to compile the project MicroSip-3.0.15.src based on pjsip. I
have added all necessary dependencies like ../lib;../ffmpeg/lib;"C:\Program
Files\Microsoft
SDKs\Windows\v6.0\Samples\Multimedia\DirectShow\BaseClasses\Debug_MBCS"

and also the libraries imm32.lib version.lib winmm.lib and
libpjproject-i386-Win32-vc8-Debug-Static.lib

But when i try to build using visual studio 2008 i have found the following
link error:

Error1error LNK2019: unresolved external symbol
_pjsua_vid_codec_set_priority referenced in function "public: void
__thiscall CmicrosipDlg::PJCreate(void)"
(?PJCreate&amp;lt; at &amp;gt;CmicrosipDlg&amp;lt; at &amp;gt;&amp;lt; at &amp;gt;QAEXXZ)microsipDlg.objmicrosip

Error2error LNK2019: unresolved external symbol
_pjsua_call_get_vid_stream_idx referenced in &lt;/pre&gt;</description>
    <dc:creator>Trent Creekmore</dc:creator>
    <dc:date>2012-05-17T16:33:00</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15560">
    <title>Re: Opus codec</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15560</link>
    <description>&lt;pre&gt;Hi Nanang,

Thanks for the point.

Actually I did a mistake integrating opus codec.
Thanks to feedback from nice guys working on opus, I got that the SDP to 
announce for opus is always opus/48000 . The codec is able to manage 
clockrate changes and re-sampling by itself.
So the problem with dynamic PT is very reduced with that :) (from 5 to 1 
:) ).

The problem I've now is that the design of pjmedia seems not made to 
manage this kind of case.
I'd like to configure the codec with pjsua_codec_set_param 
&amp;lt;http://www.pjsip.org/pjsip/docs/html/group__PJSUA__LIB__MEDIA.htm#gab5a78d3b880d47636abfa9c2077eabd0&amp;gt; 
to set the target clock rate from the application in order to avoid 
useless sampling from both pjsip.
For example, I'd like to have device open &amp;lt; at &amp;gt;16kHz, opus open &amp;lt; at &amp;gt;16kHz and 
leave opus resample and adapt to bandwidth capacity.
In pjmedia, if I keep things the way it's done for other codecs, I'll 
have device open &amp;lt; at &amp;gt;16kHz, pjmedia resample, opus open &amp;lt; at &amp;gt;48kHz (and more 
likely opus will resample again).

I th&lt;/pre&gt;</description>
    <dc:creator>Régis Montoya</dc:creator>
    <dc:date>2012-05-17T14:24:56</dc:date>
  </item>
  <item rdf:about="http://permalink.gmane.org/gmane.comp.voip.pjsip/15559">
    <title>Re: How to integrate DSP device with PJSIP</title>
    <link>http://permalink.gmane.org/gmane.comp.voip.pjsip/15559</link>
    <description>&lt;pre&gt;Robin-


The codec takes the real work, so simply call encode() and decode() functions using your DSP.  Echo-can maybe is
another function to move there at some point, if needed...

In the tightly coupled architecture that you describe (similar to an SoC), where the MIPS CPU has the IP/UDP interface
and the DSP is "attached" to the MIPS in some way (shared mem, FIFO, etc), it makes sense to only run "compute
intensive" code on the DSP, and keep everything else on the MIPS.

-Jeff


&lt;/pre&gt;</description>
    <dc:creator>Jeff Brower</dc:creator>
    <dc:date>2012-05-17T14:26:39</dc:date>
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