Query

<div><div>
<div>Hi there just want know as in case of playing&nbsp; wave file from the syntax&nbsp;</div>
<div><br></div>
<div>$ pjsua --play-file PROMPT.WAV --auto-answer

					  200 --auto-play --auto-loop \<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --max-calls 100</div>
<div><br></div>
<div>at which location PROMPT.WAV need to be placed<br>
</div>
</div></div>

query on report generation

<div><div>
<div>hi,</div>
<div>Can you guys please help me on report generations?, I want to check the performance of my call ,can i do it using PJSIP.<br>
</div>
</div></div>

RTP PT change and custom RTP PT rx/tx

Good day everyone!

I have a few questions regarding the use of pjproject.
I'm currently undecided about which version to use (1.14 or 2.0-rc)
but am tending towards the stable release branch (1.x).

These are my questions:

1. Can I switch from audio to comfort noise (change RTP payload type)?

Say I have an outgoing RTP stream and don't want to send silence but 
change the RTP payload type to CN if I don't have useful audio to send, 
how would I do it?

I did notice that there seems to be a "comfort noise generation" 
feature, but I didn't find anything on how to use it.

2. Can I send RTP packets with my own (dynamic) payload type
"on demand" using an existing stream?

Imagine I have an outgoing RTP stream on which I send real audio or 
silence (CN).
I'd like to be able to send requests (payload of the RTP packet would be 
the naked request) to the remote peer at any time using a payload type 
from the dynamic payload type range.

My expectation is that this request will be send immediately in between 
the regular RTP audio packets.

Is it possible to do this without any changes to pjproject?
If so, examples and/or pointers to the documentation would be greatly 
appreciated.

3. Can I receive RTP packets with my own payload type?

Question 2. leads to the question of how I could receive
my "RTP request packets".
The reception of such a packet should result in a function being called 
to deliver the payload and stream or call ID (to identify from which 
stream/call the packet came) to the application.

I did find the codec manager API and figured that it might to the job if 
I'd just registered my "codec" to pjsua that way. Would it work?

Regards,

Andreas

--

-- 
Dipl.-Ing. (FH) Andreas Wehrmann
Software Development
--------------------------------------------------------------
Center Communication Systems GmbH
A-1210 Wien, Ignaz-Köck-Straße 19
Sitz in Wien
FN 796 88p, Firmenbuchgericht Wien
www.centersystems.com

Tel.: +43 (0) 190 199 - 3616
Mobile: +43 (0) 664 884 75916
Fax: +43 (0) 190 199 - 2110
E-Mail: a.wehrmann <at> centersystems.com

Re: replaced calls

<div><p>Hello,<br><br>I posted a question about replaced calls on this forum several months ago, but have not got any answers back then. Our problem is that an Aastra MX-ONE user agent sends a re-invate with the replaces option for every incoming call. Even if we send back OK in our on_replace_request() python callback, the on_replaced() callback is never called. I did some debugging, and it seems that the _cb_on_call_replaced() function in pjsua.py is called, but when it tries to fetch the new call with this line<br>
&nbsp;&nbsp; new_call = self._lookup_call(new_call_id)<br>it comes up with nothing, and that is why my callback function is not called.<br><br>Anybody has any idea/suggestion about this one?<br><br>Thanks,<br>Norbert<br><br><br><br></p></div>

please tell me the location of ./pjsip-perf for perfomance bench mark

<div><div>
<div>1) Please tell me the location of ./pjsip-perf&nbsp; it is a syntax of&nbsp; performance bench mark,&nbsp; <br>
</div>
<div>2) Tell me where to place the wav file when want to play it using ($ pjsua --play-file PROMPT.WAV --auto-answer

					  200 --auto-play --auto-loop \<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --max-calls 100) syntax<br>
</div>
<div><br></div>
</div></div>

Pjsip calling landlines via a sip provider

Hi

If I configure pjsip to use a sip provider such as sipgate.co.uk is it possible to call a landline or mobile
number? And what syntax would I use?

I'm trying to use pjsip with nagios for alerting.

Thanks

PLC code

<div>
<p>Hi,</p>
<div><br></div>
<div>I was trying to walk through the pjmedia code but found that there is no packet loss concealment code. There is only a file plc_common.c but it does nothing. Can anybody help me on this. There is some plc implementation on the trunk for plc.</div>

<div><br></div>
<div>Regards,</div>
<div>Kalpesh</div>
</div>

patch for setup.py PJ_VERSION discovery

With the latest version from trunk pjsip-apps/src/python/setup.py fails
because of the new structure of pjlib/src/pj/config.c.  Find attached a
quick patch to avoid that problem.

Regards,
G.


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With the latest version from trunk pjsip-apps/src/python/setup.py fails
because of the new structure of pjlib/src/pj/config.c.  Find attached a
quick patch to avoid that problem.

Regards,
G.


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pjsua_get_transport_info for TLS

Find attached a patch to include support for TLS in
pjsua_get_transport_info function.   This is specially needed as
pjsua_get_transport_info is automatically invoked when creating the
transport using the python wrapper.

Regards,
G.


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Find attached a patch to include support for TLS in
pjsua_get_transport_info function.   This is specially needed as
pjsua_get_transport_info is automatically invoked when creating the
transport using the python wrapper.

Regards,
G.


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recepción de correo electrónico en el enlace situado más abajo.
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bad sound with pulseaudio

Hello
I'm developing a softphone under linux kubuntu 11.10, pjsip 1.12 and 
found a strange behavior.
If set softphone to use microphone via pulseaudio then other side will 
hear sound  with distortion.

my sound devices are:
0: PA [HDA Intel: ALC882 Analog (hw:0,0)] (2/8)
1: PA [HDA Intel: ALC882 Digital (hw:0,1)] (0/2)
2: PA [HDA Intel: ALC882 Analog (hw:0,2)] (2/0)
3: PA [front] (0/8)
4: PA [surround40] (0/8)
5: PA [surround41] (0/128)
6: PA [surround50] (0/128)
7: PA [surround51] (0/8)
8: PA [surround71] (0/8)
9: PA [iec958] (0/2)
10: PA [spdif] (0/2)
11: PA [pulse] (32/32)
12: PA [dmix] (0/2)
13: PA [default] (32/32)

If i set mic to [HDA Intel: ALC882 Analog (hw:0,0)] then everything is 
ok, because this is bypass pulseaudio mixer.
If set mic to [pulse] or [default], softphone will be in pulseaudio 
mixer and then sound goes with distortion.

I tested it on several computers and there was the same bug.

I also tested twinkle, another softphone that uses portuadio and there 
was not such problem, so I guess the root of problem is in portuadio 
version or in the pjsip.

--

-- 
Best regards,
Mihail Laba