Of 3.10.0 + ofmeet

Openfire Plugins openfire-meetings
1

Hi all,

Fresh install of OF 3.10.0 + OFMEET

Actually I switched from Jitsi Meet because of lack of 2 needs :

Audio-only participant + SIP participant : first was impossible, second worked but poor audio.

On OF 3.10.0 + OFMEET

Audio-only partipicipant : not working either, getting the message from Chrome “cannot take the control of Mic and Cam”

of course no cam on this one, but I thought that audio only was fine now on as announced, am I correct ?

SIP : obviously there is a bug in username trasmitted, it takes admin instead of the property in the Jitsi Videopbridge config tab.

However even when this one is fixed manually and SIP registration is OK, clicking on the DIAL icon does not make any call.

no log at all on the Sip server side.

Any Clue ?

Vince

2

sip username fault was fixed in new ofmeet plugin. try it instead. Audio only support coming soon. It is the no 1 requested feature

3

not good :

2014.12.04 22:59:03 ofmeetsip - the sip stack timer gov.nist.javax.sip.stack.timers.DefaultSipTimer has been started

2014.12.04 22:59:04 org.ifsoft.sip.SipService - VoiceBridge adding SIP registration: admin with user y1nvla host 10.0.1.140

2014.12.04 22:59:04 org.ifsoft.sip.RegisterProcessing - Start registering…10.0.1.140

2014.12.04 22:59:04 org.ifsoft.sip.RegisterProcessing - Registering with 10.0.1.140

2014.12.04 22:59:04 org.ifsoft.sip.RegisterProcessing - Could not send out the register request! Argument invalide

2014.12.04 22:59:04 org.jivesoftware.openfire.plugin.ofmeet.OfMeetPlugin - OfMeet Plugin - Initialize websockets

2014.12.04 22:59:04 org.jivesoftware.openfire.plugin.ofmeet.OfMeetPlugin - OfMeet Plugin - Initialize security

4

ok. thanks for the logs. I shall take a look

5

Dele,

First, this is a great job, you’re close to a nice platform.

I got the SIP registration work register !!!

you need to enter the IP address instead of the hostname in the fields “meetings settings / media configuration” local IP / Public IP.

I think the SIP server will not get the proper info instead.

So step 1 is good.

then 2 things :

You can actually place a call ONLY if there are already 2 participants, otherwise it transmits a confID null and then it does not dial.

when there are 2 participants, the dial button places the call BUT :

no audio either way

obviously ofmeet only offers codec PCMU but even with this on Asterisk side it does not work.

I got some messages on Asterisk side about RTP+retransmission.

Also even though call is estalished, it does not create a participant window for the SIP call in the ofmeet window.

hope this helps,

get back to me please, pretty sure it’smoving forward.

6

Hi All

I’am getting same issues. I’am able to register to asterisk box. But not able to call from web interface. I have to try with 2 or more partec.

I am agree with Vince about make it able to make also video calls.

This is a great job; if Video call will be added, will be the best…and really used platform as interoperate with also commercial devices ( like Lifesize, Cisco VCS, and so on… )

Regards

7

I made some tests and I can confirm all Vince says.

I can make audio call from web interf. ONLY if there are more then 2 person con meeting. When audio call is done no “little screen” appears on bottom right of web interf. ( that should be present to indicate that there are an “audio partecipant” to the meeting. I tested it with PROSODY+JITISTI_MEET+JIGASY and it appears )

Hope can this help as well

Regards

8

i remember that. It is an outstanding issue that you need at lease two people before it can determine who is the focus and then make the call. You should get a little icon. Please note that Openfire Meetings is using different SIP stack from Jitsi Meet, therefore jigasi component behaviour may be different with both.

9

Antonio : question : do you confirm that with 2 participant, you can make the call but NO AUDIO ?

Dele : I can make one suggestion : if you want us to help you debug more the SIP stack, you need to expose more properties

so that we can play a little but. Best is to take Jitsi sipphone, look at the settings we can adjust in a sip account + codecs, + …

and put that in properties page.

Thanks for the job again.

10

Hi Vince,

for me is happening a little bit different issue. Call is established but after some second, around 5, call is cut off.

11

make sure you are using audio mixer mode enabled

12

I enabled audio mixer mode.

I have audio from the sip phone to the webinterface but no audio from the web interface to the sipphone

on asterisk I have an alert of retramsnmission timeout 32 secs and then it cuts off.

13

restart the server for mixer mode to take effect. In translator/relay mode, the SIP client will receive audio only if it can handle multiple SSRCs

14

Does not change, it was taken into account already.

I don’t understand your point regarding translator/relay mode and multiple SSRC.

if the sip extension I call is on a sipserver (asterisk) which does not allow direct media then the RTP flows

is between Ofmeet <=> Asterisk and then Asterisk <=> Sip client.

this is a better way to handle codec transcoding when both do not use the same ones.

I will try with a sip client registered to the embedded ofmeet sipserver to see if this is better.

15

what SIP client do yo use to register with the embedded sipserver ?

tried a few ones and none is registering with the ofmeet sipserver.

I see some sip register messages hitting in the log but onclient side it times out.

Any thought ?

16

Hi Vince, I had same your tests. About client to use, I used BRIA ( on mac ) and i was able to register it face to Ofmeet sip server. I used “ngrep” to troubleshoot sip logs.

17

I use 3CX on iphone

18

Ok Eurêka.

I made I successfull call from ofmeet to a SIP client registered.

A few comments :

  • sip registration to the embedded sip server is cumbersome.

at times it works at times not you need to restart openfire.

Also you need to register with the username put in Meeting settings

but you cannot register an extension like 1000 with auth name ofmeet / pwd.

therefore when calling from ofmeet you need to call “username” and not an extension.

the restriction will be that you can have only one user.

Am I correct ?

Still not working for call to Asterisk. it registers fine but the subsequent SIP dialog does not work properly.

as said before, it rings the extension, but then no packet transmitted from ofmeet to asterisk.

19

but you cannot register an extension like 1000 with auth name ofmeet / pwd.

therefore when calling from ofmeet you need to call “username” and not an extension.

Nope. Use ofmeet/pwd as authorizing sip username/password. This is different from extension number. At least that is what I do on my 3CX sip client for iPhone. I call 1002 to join conference and web app calls 1000 to invite me.

the restriction will be that you can have only one user.

Am I correct ?

At the moment yes, because the restriction is from the user interface. The ofmeet plugin can accept many requests from the web client. I plan to fix that later.

Still not working for call to Asterisk. it registers fine but the subsequent SIP dialog does not work properly.

as said before, it rings the extension, but then no packet transmitted from ofmeet to asterisk.

Sorry about Asterisk. If you can send me your SIP logs, it could reveal something

20

Dele,

Am I wrong or since you updated jitsi meet and videobridge with OF 0.0.3

it has completed changed ?

OF entry point is now the same as jitsi meet

there is no SIP/phone icon to make a sip call

maybe I am mistaken.

thanks