Confbridge: using G.722 codec

I’m using the confbridge application to synchronize announcement aims. At the begin of an announcement, a gong soundfile is playing using the playback application.
Every connections are a SIP connection using the G722 codec.
I start an announcement from a SIP phone and it will be called two announcement aims via SIP. If the two SIP connections are established. The gong soundfile is playing.

If the soundfile is playing I hear a loud crack on one of the two loudspeaker. I think it is a transcoding problem.

I have used several configuration of the confbridge.
If I’m using the option

internal_sample_rate=auto

in the bridge profile. I hear the crack after 2,5 seconds in every announcement.

If I’m using other internal_sample_rate setting. it works better but I hear a crack in one of three attemps.

I have try other settings in the confbride configuration but nothing solve the problem.
I’m using asterisk 13.1 and the old SIP channel (chan_sip).

Any idea to solve the problem?

There are CLI tools that allow you to transcode offline, which would allow you to check the transcoding.

Hello David,
thank you for your reply.

I do not understand your answer. Which CLI tools do you mean?
If I use G711 codec everything is okay.
Is there any option which I can use?

Hello David,

I think it is not a problem of the soundfile! Because the crack is heard at different timestamps.
I think it is a timing problem during the conversion from slin format to the g722 format. Every channel needs an encoding to g722.
Is it possible, to use a soundfile format that needs a minimum of encoding?
Have anyone an idea?

I’m not conviced of that analysis, but the obvious file format to miminise transcoding issues is a signed linear one at the rate used by the destination channels, so conference bridges always have to transcode to and from signed linear in order to be able to add samples, from different source channels, together.

On the same phone every time? But never the other phone? You might try a different phone.

I always calls the same two phones! The loud crack is sometimes on that phone and sometimes on the other phone.

Let me short explain how my algorithm works.
From a SIP phone I make a call with an special number.
The dialplan calls an agi script which makes two calls to the phones. using the orginate application. If the phones establish the connection, the phones will be placed in the conference bridge.
Now, I start the announcement (gong sound file) by using the playback application.

If I’m using the g711 (alaw) codec it works fine. In case of g722 I heard a loud crack on one of the phones.

Okay I think I understand. You might try adding a short delay before the gong. Either a Wait(.2) or Playback(silence/1) might help prime the pump with a little sound in your phones’ buffers.

Thank you for this information. I will test it in short future and give you a feedback.