[pjsip] Changing codec parameter dynamically
301 at free.fr
301 at free.fr
Sat Dec 3 11:27:01 EST 2016
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
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...
2016-12-03 12:36 GMT+01:00 JOHAN LANTZ <johan.lantz at 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).
> 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 at free.fr> wrote:
>> 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?
>> Visit our blog: http://blog.pjsip.org
>> pjsip mailing list
>> pjsip at lists.pjsip.org
> Visit our blog: http://blog.pjsip.org
> pjsip mailing list
> pjsip at lists.pjsip.org
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