Changing codec parameter dynamically
Hi,
In my project where I simulate some kind of push to talk, I need to be able
to have no payload in the RTP packets when the PTT is off. I succeeded to
do that by removing the payload in a custom transport adapter and the
transport_send_rtp callback (but is it the best way to do that?).
Anyway my second need is to change the frequency of RTP packets when I
remove the payload (PTT off). For example:
with audio: RTP frame every 20ms
without audio: RTP frame every 200ms
So I was thinking of dynamically changing the ptime of the codec in use,
but I don't figure out how to do that (again this may not be the right way
to do it, any other suggestion welcome then).
Could anybody give me an hint?
Thanks.
Thibault
Hi,
You can enable VAD (voice activity detection) and when you stop talking
the stream will stop sending packets, except for periodic keepalive
packets whose rate can be controlled by some compile time constant, or
you can disable keepalive altogether.
Regards,
Bill
On 12/2/2016 6:12 PM, 301@free.fr wrote:
Hi,
In my project where I simulate some kind of push to talk, I need to be
able to have no payload in the RTP packets when the PTT is off. I
succeeded to do that by removing the payload in a custom transport
adapter and the transport_send_rtp callback (but is it the best way to
do that?).
Anyway my second need is to change the frequency of RTP packets when I
remove the payload (PTT off). For example:
with audio: RTP frame every 20ms
without audio: RTP frame every 200ms
So I was thinking of dynamically changing the ptime of the codec in
use, but I don't figure out how to do that (again this may not be the
right way to do it, any other suggestion welcome then).
Could anybody give me an hint?
Thanks.
Thibault
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
you can do it at your custom transport adapter.
for example I guess you have some application flag "without audio". when
this flag enabled, drop some rtp frames and send only 200 ms. only things
you need to do is
rewrite rtp seq and rtp timestamp correctly at your adapter if remote peer
don't like big big packet loss.
VAD work for some codec, some wouldn't because some like "SID" frames
still sending out.
On Sat, Dec 3, 2016 at 7:12 AM, 301@free.fr wrote:
Hi,
In my project where I simulate some kind of push to talk, I need to be
able to have no payload in the RTP packets when the PTT is off. I succeeded
to do that by removing the payload in a custom transport adapter and the
transport_send_rtp callback (but is it the best way to do that?).
Anyway my second need is to change the frequency of RTP packets when I
remove the payload (PTT off). For example:
with audio: RTP frame every 20ms
without audio: RTP frame every 200ms
So I was thinking of dynamically changing the ptime of the codec in use,
but I don't figure out how to do that (again this may not be the right way
to do it, any other suggestion welcome then).
Could anybody give me an hint?
Thanks.
Thibault
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Have you thought about something like:
- When you are talking or when the other peer is talking, use the sound card as always
- When there is silence, call pjsua_set_null_snd. This way you do not block the sound device for other apps needing it and you do not consume any unnecessary power. The call stays active though and the null sound maintains the timing for the rtp flow.
- When the user presses the PTT button or when you detect incoming rtp you call pjsua_set_snd_dev to activate the sound card again and so on.
- Pjsip by default does rtp and rtcp keep alive. You can configure both the content of the keep alive packets and the frequency through compiler switches easily. That way you can configure rtp keep alive to be done every 200 msec with an empty packet. (By default when you have audio, you send audio frames every 20 msec).
/Johan
From: pjsip on behalf of Gang Liu
Reply-To: pjsip list
Date: Saturday, 3 December 2016 at 04:16
To: pjsip list
Subject: Re: [pjsip] Changing codec parameter dynamically
you can do it at your custom transport adapter.
for example I guess you have some application flag "without audio". when this flag enabled, drop some rtp frames and send only 200 ms. only things you need to do is
rewrite rtp seq and rtp timestamp correctly at your adapter if remote peer don't like big big packet loss.
VAD work for some codec, some wouldn't because some like "SID" frames still sending out.
On Sat, Dec 3, 2016 at 7:12 AM, <301@free.frmailto:301@free.fr> wrote:
Hi,
In my project where I simulate some kind of push to talk, I need to be able to have no payload in the RTP packets when the PTT is off. I succeeded to do that by removing the payload in a custom transport adapter and the transport_send_rtp callback (but is it the best way to do that?).
Anyway my second need is to change the frequency of RTP packets when I remove the payload (PTT off). For example:
with audio: RTP frame every 20ms
without audio: RTP frame every 200ms
So I was thinking of dynamically changing the ptime of the codec in use, but I don't figure out how to do that (again this may not be the right way to do it, any other suggestion welcome then).
Could anybody give me an hint?
Thanks.
Thibault
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.orgmailto:pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Thank you all for your answers and proposals.
Disabling sound device seems interesting but I don't see how it can work in
my scenario (see below).
The pjsip keep-alive mechanism seems to be linked to VAD only as far as I
understand, and the compiler setting is a constraint for me because the
keep-alive period can be defined in SDP at session establishment.
To give you my scenario: I use PJSIP stack to perform "classic" telephone
calls, and in parallel to perform radio calls. Those radios calls are half
duplex; when one side doesn't transmit audio, it shall send some kind of
keep-alive packets (without payload) at a lower frequency. So when I
transmit audio, I shall receive keep-alive packets from the other side.
When the other side send me audio, I'm supposed to send keep-alive packets.
When nobody is sending audio, both sides shall send keep-alive packets.
If I disable the sound device when there is no audio on either side, I
guess this would have an impact on the other pending calls (telephone) that
the application is supporting, doesn't it? (I will loose sound on the other
calls)
When I receive audio and re-enable the sound device, then I will transmit
audio too, won't I?
What Gang Liu proposes is what I'm doing right now, but there is one
drawback: the RTP statistics are wrong this way because they still believe
that the payload has been transmitted...
I still have to find out how to retrieve the ptime used by the codec in
order to determine how many RTP frames I have to drop. Nevertheless, as for
the payload drop in the transport adapter, it seems not very efficient to
waste such processor time with things which are not useful (making RTP
frames to be dropped, making payload to be removed...).
But at least it works :-) I will see the performance impact with large
number of calls...
Thibault
2016-12-03 12:36 GMT+01:00 JOHAN LANTZ johan.lantz@telefonica.com:
Have you thought about something like:
- When you are talking or when the other peer is talking, use the
sound card as always
- When there is silence, call pjsua_set_null_snd. This way you do not
block the sound device for other apps needing it and you do not consume any
unnecessary power. The call stays active though and the null sound
maintains the timing for the rtp flow.
- When the user presses the PTT button or when you detect incoming rtp
you call pjsua_set_snd_dev to activate the sound card again and so on.
- Pjsip by default does rtp and rtcp keep alive. You can configure
both the content of the keep alive packets and the frequency through
compiler switches easily. That way you can configure rtp keep alive to be
done every 200 msec with an empty packet. (By default when you have audio,
you send audio frames every 20 msec).
/Johan
From: pjsip on behalf of Gang Liu
Reply-To: pjsip list
Date: Saturday, 3 December 2016 at 04:16
To: pjsip list
Subject: Re: [pjsip] Changing codec parameter dynamically
you can do it at your custom transport adapter.
for example I guess you have some application flag "without audio". when
this flag enabled, drop some rtp frames and send only 200 ms. only things
you need to do is
rewrite rtp seq and rtp timestamp correctly at your adapter if remote peer
don't like big big packet loss.
VAD work for some codec, some wouldn't because some like "SID" frames
still sending out.
On Sat, Dec 3, 2016 at 7:12 AM, 301@free.fr wrote:
Hi,
In my project where I simulate some kind of push to talk, I need to be
able to have no payload in the RTP packets when the PTT is off. I succeeded
to do that by removing the payload in a custom transport adapter and the
transport_send_rtp callback (but is it the best way to do that?).
Anyway my second need is to change the frequency of RTP packets when I
remove the payload (PTT off). For example:
with audio: RTP frame every 20ms
without audio: RTP frame every 200ms
So I was thinking of dynamically changing the ptime of the codec in use,
but I don't figure out how to do that (again this may not be the right way
to do it, any other suggestion welcome then).
Could anybody give me an hint?
Thanks.
Thibault
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org