manjeet | 1 Apr 09:23 2009
Picon

Re: transport_send_rtcp() segfault issue (Johan Lantz)

Thanks Johan.


--- On Tue, 3/31/09, Johan Lantz <johan.lantz <at> genaker.net> wrote:
From: Johan Lantz <johan.lantz <at> genaker.net>
Subject: Re: [pjsip] transport_send_rtcp() segfault issue (Johan Lantz)
To: pjsip <at> lists.pjsip.org
Date: Tuesday, March 31, 2009, 1:48 PM

Hi Manjeet

A simpler approach could perhaps be to not use the heap at all. If you are
sending RTCP APP the max size is probably quite small.


//
pj_uint8_t TXBuffer[MAX_TX_PACKET_SIZE];
pj_uint8_t actualPacketSize;

my_create_pkt_function(TXBuffer, &amp ;actualPacketSize);

transport_send_rtcp(tp, TXBuffer, actualPacketSize);
//

Now when transport returns you do not have to care about freeing the memory at
all.

/Johan

Message: 5
Date: Mon, 30 Mar 2009 04:50:28 -0700 (PDT)
From: manjeet <manjeetss1 <at> yahoo.com>
Subject: [pjsip] transport_send_rtcp() segfault issue
To: pjsip <at> lists.pjsip.org
Message-ID: <602003.49962.qm <at> web52009.mail.re2.yahoo.com>
Content-Type: text/plain; charset="us-ascii"

Hi,

I am using "transport_send_rtcp()" to send the RTCP APP packet to the
network.

The signature is as below :- transport_send_rtcp(pjmedia_transport *tp, const
void *pkt, pj_size_t size);

Now i am passing a "pkt", which is allocated on the heap in my
application as the 2nd parameter to "transport_send_rtcp()".

Question is do i need to "delete pkt" in my application once the
transport_send_rtcp() return PJ_SUCC ESS, or transport_send_rtcp() itseld will
take care of this?

I tried deleting it when transport_send_rtcp() returns PJ_SUCCESS, but it
segfaults sometimes.

Any help will be appreciated?

Thanks.


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<div>
<table cellspacing="0" cellpadding="0" border="0"><tr><td valign="top">Thanks Johan.<br><br><br>--- On Tue, 3/31/09, Johan Lantz &lt;johan.lantz <at> genaker.net&gt; wrote:<br><blockquote>From: Johan Lantz &lt;johan.lantz <at> genaker.net&gt;<br>Subject: Re: [pjsip] transport_send_rtcp() segfault issue (Johan Lantz)<br>To: pjsip <at> lists.pjsip.org<br>Date: Tuesday, March 31, 2009, 1:48 PM<br><br>Hi Manjeet<br><br>A simpler approach could perhaps be to not use the heap at all. If you are<br>sending RTCP APP the max size is probably quite small.<br><br><br>//<br>pj_uint8_t TXBuffer[MAX_TX_PACKET_SIZE];<br>pj_uint8_t actualPacketSize;<br><br>my_create_pkt_function(TXBuffer, &amp;amp
 ;actualPacketSize);<br><br>transport_send_rtcp(tp, TXBuffer, actualPacketSize);<br>//<br><br>Now when transport returns you do not have to care about freeing the memory
 at<br>all.<br><br>/Johan <br><br>Message: 5<br>Date: Mon, 30 Mar 2009 04:50:28 -0700 (PDT)<br>From: manjeet &lt;manjeetss1 <at> yahoo.com&gt;<br>Subject: [pjsip] transport_send_rtcp() segfault issue<br>To: pjsip <at> lists.pjsip.org<br>Message-ID: &lt;602003.49962.qm <at> web52009.mail.re2.yahoo.com&gt;<br>Content-Type: text/plain; charset="us-ascii"<br><br>Hi,<br><br>I am using "transport_send_rtcp()" to send the RTCP APP packet to the<br>network.<br><br>The signature is as below :- transport_send_rtcp(pjmedia_transport *tp, const<br>void *pkt, pj_size_t size);<br><br>Now i am passing a "pkt", which is allocated on the heap in my<br>application as the 2nd parameter to "transport_send_rtcp()".<br><br>Question is do i need to "delete pkt" in my application once the<br>transport_send_rtcp() return PJ_SUCC
 ESS, or transport_send_rtcp() itseld will<br>take care of this?<br><br>I tried deleting it when transport_send_rtcp() returns PJ_SUCCESS, but it<br>segfaults
 sometimes.<br><br>Any help will be appreciated?<br><br>Thanks.<br><br><br>_______________________________________________<br>Visit our blog: http://blog.pjsip.org<br><br>pjsip mailing list<br>pjsip <at> lists.pjsip.org<br>http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org<br>
</blockquote>
</td></tr></table>
<br>
</div>
Soh Wei Sin | 1 Apr 12:36 2009
Picon

Can anyone help with problem of PJSUA unregister?

Hi all,
I have a problem where I try to modified the PJProject 1.0.1 pjsua for support for  IPV6 register. However, whenever the Client that use sip id of "sip:ims <at> [2001:db8:121:1:3]" register to SER 0.9.6, it will straight away unregister with the error that:
11:52:26.376    pjsua_acc.c  Unregistration sent
11:52:26.376      sip_reg.c  Invalid Contact URI: "<sip:ims <at> 2001:DB8:121:0:0:0:0:3:6060;transport=UDP>"
11:52:26.376    pjsua_acc.c  Client registration initialization error: Invalid URI (PJSIP_EINVALIDURI) [status=171039]
11:52:26.376    pjsua_acc.c  Unable to create registration: Invalid URI (PJSIP_EINVALIDURI) [status=171039]
11:52:26.376  pjsua_core.c  RX 630 bytes Response msg 200/REGISTER/cseq=60649 (rdata0x81d8824) from UDP 2001:db8:121::3:5060

I'm urgently need solve this problem but seem nobody reply for my question. Hope anyone can help me.

Thanks in advance.

Regards,
ws

<div>
<div><div>Hi all,<br>I have a problem where I try to modified the PJProject 1.0.1 pjsua for support for&nbsp; IPV6 register. However, whenever the Client that use sip id of "sip:ims <at> [2001:db8:121:1:3]" register to SER 0.9.6, it will straight away unregister with the error that:<br>11:52:26.376&nbsp; &nbsp; pjsua_acc.c&nbsp; Unregistration sent<br> 11:52:26.376&nbsp; &nbsp; &nbsp; sip_reg.c&nbsp; Invalid Contact URI: "&lt;sip:ims <at> 2001:DB8:121:0:0:0:0:3:6060;transport=UDP&gt;"<br> 11:52:26.376&nbsp; &nbsp; pjsua_acc.c&nbsp; Client registration initialization error: Invalid URI (PJSIP_EINVALIDURI) [status=171039]<br> 11:52:26.376&nbsp; &nbsp; pjsua_acc.c&nbsp; Unable to create registration: Invalid URI (PJSIP_EINVALIDURI) [status=171039]<br> 11:52:26.376&nbsp;  pjsua_core.c&nbsp;
 RX 630 bytes Response msg 200/REGISTER/cseq=60649 (rdata0x81d8824) from UDP 2001:db8:121::3:5060<br><br>I'm urgently need solve this problem but seem nobody reply for my question. Hope anyone can help me.<br><br>Thanks in advance.<br><br>Regards,<br>ws<br>
</div></div>
<br>
</div>
Benny Prijono | 1 Apr 12:49 2009

Re: Regarding porting PJSIP to a platform with builtin HW codec

2009/3/31 Wang Eric <eric_wanga <at> hotmail.com>

To whom may concerned,
 
Currently, I'm going to porting PJSIP to a specified platform with built in HW codec.
AFter a glance with the PJSIP release V1.1. I still get the following problems.
(1) To enable PJ_CONFIG_NOKIA_APS_DIRECT
Most of the dependence are at "pjmedia" and "pjsua-lib", is it correct?


I'm not sure what you mean by dependence, but most modifications that we did was in pjmedia and pjsua-lib indeed. You don't need to modify pjmedia/pjsua-lib/or any pj libs in order to use APS-Direct, please see http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct of course.
 

(2) What is the major to-do job for me?
Enable PJ_CONFIG_NOKIA_APS_DIRECT and replace pjsipua by our own module. (High layer apps and call logic control)
Ignore the symbian_ua, symbian_ua_gui folder under "pjsip-apps" folder
 

No no no. Your "only" major job is to implement your own sound device abstraction for your hardware, according to the new Audio Device API spec (http://trac.pjsip.org/repos/wiki/Audio_Dev_API). Then enable "APS-Direct mode" when building the libs, following the wiki link above, or by following sample configs provided by include/pj/config_site_sample.h (there's even a sample config there to enable "APS-Direct" mode with WMME, though this is experimental).

Your application will be able to use pjsua-lib as usual, minus the conferencing/mixing feature of course. For more info please see the Wiki links above.

cheers
 Benny

 

Thanks for your help in advance!

下載 Windows Live Messenger 9.0,多元溝通、盡情分享,和即時通訊好友線上同樂!— 立即下載

_______________________________________________
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<div>
<p>2009/3/31 Wang Eric <span dir="ltr">&lt;<a href="mailto:eric_wanga <at> hotmail.com">eric_wanga <at> hotmail.com</a>&gt;</span><br></p>
<div class="gmail_quote">
<blockquote class="gmail_quote">

<div>
To whom may concerned,<br>
&nbsp;<br>
Currently, I'm going to porting PJSIP to a specified platform with built in HW codec.<br>AFter a glance with the PJSIP release V1.1. I still get the following problems.<br>
(1) To enable PJ_CONFIG_NOKIA_APS_DIRECT<br>Most of the dependence are at "pjmedia" and "pjsua-lib", is it correct?<br><br>
</div>
</blockquote>
<div>
<br>I'm not sure what you mean by dependence, but most modifications that we did was in pjmedia and pjsua-lib indeed. You don't need to modify pjmedia/pjsua-lib/or any pj libs in order to use APS-Direct, please see <a href="http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct">http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct</a> of course.<br>
&nbsp;<br><br>
</div>
<blockquote class="gmail_quote"><div>(2) What is the major to-do job for me?<br>Enable PJ_CONFIG_NOKIA_APS_DIRECT and replace pjsipua by our own module. (High layer apps and call logic control)<br>
Ignore the symbian_ua, symbian_ua_gui folder under "pjsip-apps" folder<br>
&nbsp;<br>
</div></blockquote>
<div>
<br>No no no. Your "only" major job is to implement your own sound device abstraction for your hardware, according to the new Audio Device API spec (<a href="http://trac.pjsip.org/repos/wiki/Audio_Dev_API">http://trac.pjsip.org/repos/wiki/Audio_Dev_API</a>). Then enable "APS-Direct mode" when building the libs, following the wiki link above, or by following sample configs provided by include/pj/config_site_sample.h (there's even a sample config there to enable "APS-Direct" mode with WMME, though this is experimental).<br><br>Your application will be able to use pjsua-lib as usual, minus the conferencing/mixing feature of course. For more info please see the Wiki links above.<br><br>cheers<br>&nbsp;Benny<br><br>&nbsp;<br><br>
</div>
<blockquote class="gmail_quote">
<div>
Thanks for your help in advance!<br><br>&#19979;&#36617; Windows Live Messenger 9.0&#65292;&#22810;&#20803;&#28317;&#36890;&#12289;&#30433;&#24773;&#20998;&#20139;&#65292;&#21644;&#21363;&#26178;&#36890;&#35338;&#22909;&#21451;&#32218;&#19978;&#21516;&#27138;&#65281;&mdash; <a href="http://download.live.com/messenger" target="_blank">&#31435;&#21363;&#19979;&#36617;</a>
</div>
<br>_______________________________________________<br>
Visit our blog: <a href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a><br><br>
pjsip mailing list<br><a href="mailto:pjsip <at> lists.pjsip.org">pjsip <at> lists.pjsip.org</a><br><a href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a><br><br>
</blockquote>
</div>
<br>
</div>
Benny Prijono | 1 Apr 12:51 2009

Re: How to handle caller get BYE when on early dialog

Hi Gang,

Yeah I still have your patch pending for review/integration. Please bear with me (it will take a while!). :)

cheers
 Benny

2009/3/30 Gang Liu <gangban.lau <at> gmail.com>
Dear benny,
        I know callee's UA MUST not send a BYE on early dialog. But pls consider below msg flow:

1,  callee get INVITE from caller, and respond 200 OK first time(this packet was lost);
    
2,  callee use pjsip_inv_end_session() to disconnect the call. BYE will go to caller because callee is in PJSIP_INV_STATE_CONNECTING state.

3,  then caller get BYE on early dialog.

      There are no logic to handel this at function inv_on_state_early().
      I tried to use inv_respond_incoming_bye() to create a respond for BYE. But callee still create ACK request for retransmission 200 OK after dialog in disconnected state. And the pool used by ACK tdata still there after long time.
      And I also found there was transacton/tata pool leak when dumping pj_caching_pool, but related dialog not there. Is there anyway to release these unused pools on the fly?
  
+    } else if (inv->role == PJSIP_ROLE_UAC &&
+           tsx->role == PJSIP_ROLE_UAS &&
+           tsx->method.id == PJSIP_BYE_METHOD &&
+           tsx->state < PJSIP_TSX_STATE_COMPLETED &&
+           e->body.tsx_state.type == PJSIP_EVENT_RX_MSG )
+    {
+       /*
+        * Handle incoming BYE request.
+        */
+
+       inv_respond_incoming_bye(inv, tsx, e->body.tsx_state.src.rdata, e);
+
     }

regards,
Gang 


_______________________________________________
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<div>
<p>Hi Gang,<br><br>Yeah I still have your patch pending for review/integration. Please bear with me (it will take a while!). :)<br><br>cheers<br>&nbsp;Benny<br><br></p>
<div class="gmail_quote">2009/3/30 Gang Liu <span dir="ltr">&lt;<a href="mailto:gangban.lau <at> gmail.com">gangban.lau <at> gmail.com</a>&gt;</span><br><blockquote class="gmail_quote">Dear benny,<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I know callee's UA MUST not send a BYE on early dialog. But pls consider below msg flow:<br><br>1,&nbsp; callee get INVITE from caller, and respond 200 OK first time(this packet was lost);<br>&nbsp;&nbsp;&nbsp;&nbsp; <br>
2,&nbsp; callee use pjsip_inv_end_session() to disconnect the call. BYE will go to caller because callee is in PJSIP_INV_STATE_CONNECTING state.<br><br>3,&nbsp; then caller get BYE on early dialog.<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; There are no logic to handel this at function inv_on_state_early().<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I tried to use inv_respond_incoming_bye() to create a respond for BYE. But callee still create ACK request for retransmission 200 OK after dialog in disconnected state. And the pool used by ACK tdata still there after long time.<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; And I also found there was transacton/tata pool leak when dumping pj_caching_pool, but related dialog not there. Is there anyway to release these unused pools on the fly?<br>&nbsp;&nbsp; <br>+&nbsp;&nbsp;&nbsp; } else if (inv-&gt;role == PJSIP_ROLE_UAC &amp;&amp;<br>

+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; tsx-&gt;role == PJSIP_ROLE_UAS &amp;&amp;<br>+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; tsx-&gt;<a href="http://method.id" target="_blank">method.id</a> == PJSIP_BYE_METHOD &amp;&amp;<br>+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; tsx-&gt;state &lt; PJSIP_TSX_STATE_COMPLETED &amp;&amp;<br>

+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; e-&gt;body.tsx_state.type == PJSIP_EVENT_RX_MSG )<br>+&nbsp;&nbsp;&nbsp; {<br>+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; /*<br>+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; * Handle incoming BYE request.<br>+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; */<br>+<br>+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; inv_respond_incoming_bye(inv, tsx, e-&gt;body.tsx_state.src.rdata, e);<br>

+<br>&nbsp;&nbsp;&nbsp;&nbsp; } <br><br>regards,<br>Gang&nbsp; <br><br><br>_______________________________________________<br>
Visit our blog: <a href="http://blog.pjsip.org" target="_blank">http://blog.pjsip.org</a><br><br>
pjsip mailing list<br><a href="mailto:pjsip <at> lists.pjsip.org">pjsip <at> lists.pjsip.org</a><br><a href="http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org" target="_blank">http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org</a><br><br>
</blockquote>
</div>
<br>
</div>
Kapil Pendse | 1 Apr 14:56 2009
Picon

iLBC + PJSUA on TI DaVinci

Hello,


I've compiled the pjproject 1.0.1 for TI DaVinci DM355. Initially there was problem in the audio. Everything was jittery and nothing could be made out. Then I used the following method to compile:

$ ./aconfigure --host=armv5tl-montavista-linuxeabi CFLAGS='-O2 -msoft-float -DNDEBUG' --disable-floating-point
$ make ARCH=arm CROSS_COMPILE=arm_v5t_le-

And executed pjsua as:

$ ./pjsua-armv5tl-montavista-linux-gnueabi --app-log-level=2 --stereo --ec-tail=0 --add-codec pcmu --dis-codec speex --dis-codec ilbc --dis-codec GSM --dis-codec G722 --clock-rate=8000 --snd-clock-rate=8000

Then the audio was clean. Since --add-codec pcmu was used, PCMU codec was used by default.

Then I changed to iLBC by running the same pjsua build as:

$ ./pjsua-armv5tl-montavista-linux-gnueabi --app-log-level=2 --stereo --ec-tail=0 --dis-codec pcmu --dis-codec speex --add-codec ilbc --dis-codec GSM --dis-codec G722 --clock-rate=8000 --snd-clock-rate=8000

Now again the audio is very jittery, just like the first problem I had faced (described above). I'm not sure if "jittery" is the correct word to describe it. I can barely make out that someone is speaking from the other end of the call, but cannot understand what it is... I'm new to VOIP and PJSIP so kindly help me out.

I even changed the default iLBC mode from 30 to 20:

pjmedia/src/pjmedia-codec/ilbc.c:
//#define DEFAULT_MODE 30
#define DEFAULT_MODE 20

pjsip/include/pjsua-lib/pjsua.h
#ifndef PJSUA_DEFAULT_ILBC_MODE
//#   define PJSUA_DEFAULT_ILBC_MODE 30
#   define PJSUA_DEFAULT_ILBC_MODE 20
#endif

I also played around with the --ec-tail=X command line parameter. There is some change in way the received audio plays out but it is not at all understandable.

Kindly help me out.

Best regards,
Kapil Pendse
TekKraft


--
"The Power to Imagine, is The Power to Create!"
-TTux
<div>
<p>Hello,</p>
<div><br></div>
<div>I've compiled the pjproject 1.0.1 for TI DaVinci DM355. Initially there was problem in the audio. Everything was jittery and nothing could be made out. Then I used the following method to compile:</div>
<div><br></div>
<div>
<div>$ ./aconfigure --host=armv5tl-montavista-linuxeabi CFLAGS='-O2 -msoft-float -DNDEBUG' --disable-floating-point</div>
<div>$ make ARCH=arm CROSS_COMPILE=arm_v5t_le-</div>
<div><br></div>
<div>
And executed pjsua as:</div>
<div>
<div><br></div>
<div>$ ./pjsua-armv5tl-montavista-linux-gnueabi --app-log-level=2 --stereo --ec-tail=0 --add-codec pcmu --dis-codec speex --dis-codec ilbc --dis-codec GSM --dis-codec G722 --clock-rate=8000 --snd-clock-rate=8000</div>
<div><br></div>
</div>
<div>Then the audio was clean. Since --add-codec pcmu was used, PCMU codec was used by default.</div>
<div><br></div>
<div>Then I changed to iLBC by running the same pjsua build as:</div>
<div><br></div>
<div><div>
<div>$ ./pjsua-armv5tl-montavista-linux-gnueabi --app-log-level=2 --stereo --ec-tail=0 --dis-codec pcmu --dis-codec speex --add-codec ilbc --dis-codec GSM --dis-codec G722 --clock-rate=8000 --snd-clock-rate=8000</div>
<div><br></div>
<div>Now again the audio is very jittery, just like the first problem I had faced (described above). I'm not sure if "jittery" is the correct word to describe it. I can barely make out that someone is speaking from the other end of the call, but cannot understand what it is... I'm new to VOIP and PJSIP so kindly help me out.</div>
</div></div>
<div><br></div>
<div>I even changed&nbsp;the&nbsp;default iLBC mode from 30 to 20:</div>
<div><br></div>
<div>pjmedia/src/pjmedia-codec/ilbc.c:</div>
<div>//#define DEFAULT_MODE<span class="Apple-tab-span">	</span>30<br>
</div>
<div>#define DEFAULT_MODE<span class="Apple-tab-span">	</span>20<br>
</div>
<div><br></div>
<div>pjsip/include/pjsua-lib/pjsua.h<br>
</div>
<div>
<div>#ifndef PJSUA_DEFAULT_ILBC_MODE</div>
<div>//# &nbsp; define PJSUA_DEFAULT_ILBC_MODE<span class="Apple-tab-span">	</span>30</div>
<div># &nbsp; define PJSUA_DEFAULT_ILBC_MODE<span class="Apple-tab-span">	</span>20</div>
<div>#endif</div>
<div><br></div>
</div>
<div>I also played around with the --ec-tail=X command line parameter. There is some change in way the received audio plays out but it is not at all understandable.</div>
<div><br></div>
<div>Kindly help me out.</div>
<div><br></div>
<div>Best regards,</div>
<div>Kapil Pendse</div>
<div>TekKraft</div>
<div><br></div>
<div><br></div>-- <br>"The Power to Imagine, is The Power to Create!"<br>
-TTux<br>
</div>
</div>
Ruud Klaver | 1 Apr 15:02 2009

Possible race conditions when destroying wav_player port

Hi,

I've received reports of the following assertion failure:
../src/pjmedia/wav_player.c:554:
file_get_frame: Assertion `fport->fmt_tag == PJMEDIA_WAVE_FMT_TAG_ULAW
|| fport->fmt_tag == PJMEDIA_WAVE_FMT_TAG_ALAW'
failed

file_get_frame seems to be called within the context of the soundcard  
thread (which makes sense). What I think is happening is that I'm  
calling pjmedia_port_destroy() on the port and  
pjsip_endpt_release_pool() on the pool  containing it while the  
soundcard thread is in this function.

My questions are:
1) Is this possible?
2) If so, what can I do to prevent this?

Ruud Klaver
AG Projects

Athar Shiraz Siddiqui | 1 Apr 21:02 2009
Picon

Question about PJSIP

So we were reading this: " Very small footprint  With less than 150KB
for complete SIP features"
We needed just the register, make call and answer call features. Where
do we find this file/ code/dependencies with < 150KB for our embedded
project?

--

-- 
Shiraz

500 Riverside Drive
#425
New York, NY 10027
(703) 879-8342 (skype prefer)
(571) 276 2404 (cell)
(212) 316 8630 (landline) (extn 8630)

Michael | 1 Apr 22:01 2009
Picon

Threads question

Hi All,

I’m trying to use threads in my SIP driver under windows but without experience with threads  I’m only
receiving: "Calling pjlib from unknown/external thread. You must register external threads with
pj_thread_register() before calling any pjlib functions." or it crashes as soon as it reaches
pj_pool_create and I can’t do anything.

I’m trying use simple PJSUA in my method responsible for starting SIP communication and from what I’ve
read I have to register pjsip thread. I would be very thankful for any advice when and where can I (have to)
register a thread, call pj_pool_create, pj_thread_create etc, or do I have to consider other issues? .
Maybe someone have sample code with threads, besides those from pjsip site?.

Thanks in advance
Michael

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Athar Shiraz Siddiqui | 2 Apr 00:11 2009
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Tutorial Making VOIP Call Using SIP

We were looking through SIP and we would like to have the minimal code
possible to register, make a call and answer a call.
We would like to not use any unnecessary libraries or files.
1) Is there a place where we can start to take out such unnecessary code?
We are looking at this :
http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm
This is ideal but this has pjmedia related dependencies (in pjsua.h
which has pj-mediacodecs refs). If this was a self sufficient package
which just handles this stripped down functionality that would be
great.

2) we would like to avoid using third party libraries completely and
just use our own c code to make calls, answer and register. Are there
some pointers or hints in pjsip or some resource on the web?

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ljmscsq | 2 Apr 04:16 2009

How to establish multicast communication using pjmedia functions?

Hi,I am making a project and it will use multicast technology. I know pjmedia can be used to facilitate RTP/RTCP unicast or multicast communication. I have found a simple example(http://www.pjsip.org/pjmedia/docs/html/page_pjmedia_samples_streamutil_c.htm) and it uses unicast communication . Can I change it into multicast communication easily ? How to open multicast communication instead of unicast one?
Thank you in advance



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<div>
<div>Hi,I am making a project and it will use multicast technology. I know pjmedia can be used to facilitate RTP/RTCP unicast or multicast communication. I have found a simple example(<a href="http://www.pjsip.org/pjmedia/docs/html/page_pjmedia_samples_streamutil_c.htm">http://www.pjsip.org/pjmedia/docs/html/page_pjmedia_samples_streamutil_c.htm</a>) and it uses unicast communication .&nbsp;Can I&nbsp;change it into multicast communication easily ? How to open multicast communication instead of unicast one? </div>
<div>Thank you in advance<br><br>
</div>
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