[jitsi-dev] Configuration for XMPP/Webrtc side to set RTP translator mode in Jigasi


#1

Hi,

After changing the USE_TRANSLATOR_IN_CONFERENCE to true in
sip-communicator.properties file, Jigasi still works in mixer mode.

It looks like we also need to change the xmpp/webrtc side to work in
translator mode instead of mixer, however there is no such configuration
for xmpp/webrtc side.

I made the change in the service/protocol/Call.java as below marked by ^^^

        useTranslator
            = accountID.getAccountPropertyBoolean(
                    ProtocolProviderFactory.USE_TRANSLATOR_IN_CONFERENCE,
                    true);
                    ^^^^^

From the wireshark capture between Jigasi and FreeSwitch, I can see Jigasi

sends two audio streams to FreeSwitch.

Damin/Emil, is my understanding correct, and what is the better way to set
XMPP/webrtc side to work in RTP translator mode?

Thanks,

/Kaiduan


#2

Hi,

There is a specific logic in jigasi for overriding/adding properties
used for the xmpp account. All you need is to add the following
property:
org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true

Check out the comment here:
https://github.com/jitsi/jigasi/blob/master/jigasi-home/sip-communicator.properties#L49

Regards
damencho

···

On Mon, Apr 10, 2017 at 4:05 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:

Hi,

After changing the USE_TRANSLATOR_IN_CONFERENCE to true in
sip-communicator.properties file, Jigasi still works in mixer mode.

It looks like we also need to change the xmpp/webrtc side to work in
translator mode instead of mixer, however there is no such configuration for
xmpp/webrtc side.

I made the change in the service/protocol/Call.java as below marked by ^^^

        useTranslator
            = accountID.getAccountPropertyBoolean(
                    ProtocolProviderFactory.USE_TRANSLATOR_IN_CONFERENCE,
                    true);
                    ^^^^^

From the wireshark capture between Jigasi and FreeSwitch, I can see Jigasi
sends two audio streams to FreeSwitch.

Damin/Emil, is my understanding correct, and what is the better way to set
XMPP/webrtc side to work in RTP translator mode?

Thanks,

/Kaiduan

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#3

After adding the org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true,
it works as expected.

Many thanks Damencho.

/Kaiduan

···

On Mon, Apr 10, 2017 at 5:32 PM, Damian Minkov <damencho@jitsi.org> wrote:

Hi,

There is a specific logic in jigasi for overriding/adding properties
used for the xmpp account. All you need is to add the following
property:
org.jitsi.jigasi.xmpp.acc.USE_TRANSLATOR_IN_CONFERENCE=true

Check out the comment here:
https://github.com/jitsi/jigasi/blob/master/jigasi-home/sip-communicator.
properties#L49

Regards
damencho

On Mon, Apr 10, 2017 at 4:05 PM, Kaiduan Xie <kaiduanx@gmail.com> wrote:
> Hi,
>
> After changing the USE_TRANSLATOR_IN_CONFERENCE to true in
> sip-communicator.properties file, Jigasi still works in mixer mode.
>
> It looks like we also need to change the xmpp/webrtc side to work in
> translator mode instead of mixer, however there is no such configuration
for
> xmpp/webrtc side.
>
> I made the change in the service/protocol/Call.java as below marked by
^^^
>
> useTranslator
> = accountID.getAccountPropertyBoolean(
> ProtocolProviderFactory.USE_
TRANSLATOR_IN_CONFERENCE,
> true);
> ^^^^^
>
> From the wireshark capture between Jigasi and FreeSwitch, I can see
Jigasi
> sends two audio streams to FreeSwitch.
>
> Damin/Emil, is my understanding correct, and what is the better way to
set
> XMPP/webrtc side to work in RTP translator mode?
>
> Thanks,
>
> /Kaiduan
>
> _______________________________________________
> dev mailing list
> dev@jitsi.org
> Unsubscribe instructions and other list options:
> http://lists.jitsi.org/mailman/listinfo/dev

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