Hi all,
We try to build a VoIP application for simulating several audio media
features on top of the pjsip stack. This application should have the
following features related to the pjsip-stack:
In my opinion there are two different approaches to a solution, every one
leads to several poblems I want to describe in relation to the model used.
I guess, that a solution would require significant changes/extensions
within the pjsip stack.
*** Recording model (suitable in both models): ***
Jack client x input port - conferenc bridge master port - tx to all
transport streams (that's already the case)
*** Playback models: ***
Model 1
---====
Jack client 1 output port - conferenc bridge - rx transport stream 1
Jack client 2 output port - conferenc bridge - rx transport stream 2
Jack client 3 output port - conferenc bridge - rx transport stream 3
...
Jack client n-1 output port - conferenc bridge - rx transport stream n-1
Issues:
Solution:
Model 2
---====
Jack client x output port 1 - conferenc bridge - rx transport stream 1
Jack client x output port 2 - conferenc bridge - rx transport stream 2
Jack client x output port 3 - conferenc bridge - rx transport stream 3
...
Jack client x output port n-1 - conferenc bridge - rx transport stream n-1
Issues:
Starting from Jack I retraced the output port/channel information to the
conference bridge. I figured out that the information about multiple
ports/channels of a sounddevice gets lost in the transition from portaudio
(functions NonAdaptingProcess/AdaptingOutputProcess) to internal
functions like PaPlayerCallback. It looks like only the outputbuffers of the
first port/channel will be processed within pjspip . (variable void *output,
array which holds per channel output-buffers on the PortAudio-side)
Solution:
Any thoughts about that, did anybody something similar before?
--
Thanks And Regards,
Thomas Kluge
Hi Thomas,
Sorry for the late reply.
If I read your post correctly, I think you're modeling the solution based on
the assumption that we can't use multiple sound devices with the conference
bridge. That's not true actually.
You can create another sound device (assuming PA Jack supports this), create
and connect a splitcomb to the snd dev, then create a rev_port from the
splitcomb, and register this rev_port to the bridge (just as you register
any other port to the bridge). There is a snippet for this in pjsua_app.c,
see the stereo_demo() function towards the bottom of the file.
Alternatively, since you don't need mixing, why don't you just disable the
bridge altogether, and connect the stream directly to the corresponding jack
sound dev? This is explained here:
http://trac.pjsip.org/repos/wiki/FAQ#pjsua-lib-perf
cheers
Benny
On Wed, Jan 21, 2009 at 10:28 AM, Thomas Kluge T.Kluge@gmx.com wrote:
Hi all,
We try to build a VoIP application for simulating several audio media
features on top of the pjsip stack. This application should have the
following features related to the pjsip-stack:
In my opinion there are two different approaches to a solution, every one
leads to several poblems I want to describe in relation to the model used.
I guess, that a solution would require significant changes/extensions
within the pjsip stack.
*** Recording model (suitable in both models): ***
Jack client x input port - conferenc bridge master port - tx to all
transport streams (that's already the case)
*** Playback models: ***
Model 1
---====
Jack client 1 output port - conferenc bridge - rx transport stream 1
Jack client 2 output port - conferenc bridge - rx transport stream 2
Jack client 3 output port - conferenc bridge - rx transport stream 3
...
Jack client n-1 output port - conferenc bridge - rx transport stream n-1
Issues:
Solution:
Model 2
---====
Jack client x output port 1 - conferenc bridge - rx transport stream 1
Jack client x output port 2 - conferenc bridge - rx transport stream 2
Jack client x output port 3 - conferenc bridge - rx transport stream 3
...
Jack client x output port n-1 - conferenc bridge - rx transport stream n-1
Issues:
Starting from Jack I retraced the output port/channel information to the
conference bridge. I figured out that the information about multiple
ports/channels of a sounddevice gets lost in the transition from portaudio
(functions NonAdaptingProcess/AdaptingOutputProcess) to internal
functions like PaPlayerCallback. It looks like only the outputbuffers of
the
first port/channel will be processed within pjspip . (variable void
*output,
array which holds per channel output-buffers on the PortAudio-side)
Solution:
Any thoughts about that, did anybody something similar before?
--
Thanks And Regards,
Thomas Kluge
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Hi,
I hope you could help me here, I have 2 questions:
Did anyone implement the RTCP-XR in PJSUA API?
In the documentation it says: "PJMEDIA implements subsets of RTCP XR
specification"
Thanks!
Noga
On Thu, Jan 22, 2009 at 6:46 AM, Noga Yehudai nogab@juniper.net wrote:
Hi,
I hope you could help me here, I have 2 questions:
– What were the parameters that were implemented, and what was left out
http://trac.pjsip.org/repos/ticket/513
cheers
Benny
Thanks!
Noga
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Benny hi,
Thanks for the prompt response!
Noga
From: pjsip-bounces@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Benny Prijono
Sent: Thursday, January 22, 2009 10:10 AM
To: pjsip list
Subject: Re: [pjsip] RTCP-XR
On Thu, Jan 22, 2009 at 6:46 AM, Noga Yehudai nogab@juniper.net wrote:
Hi,
I hope you could help me here, I have 2 questions:
1. Did anyone implement the RTCP-XR in PJSUA API?
Yes: http://www.pjsip.org/pjmedia/docs/html/group__PJMED__RTCP__XR.htm
2. In the documentation it says: "PJMEDIA implements subsets of
RTCP XR specification"
- What were the parameters that were implemented, and what was
left out
http://trac.pjsip.org/repos/ticket/513
cheers
Benny
Thanks!
Noga
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