[jitsi-users] SIP RTP Port Range


#1

Let me see if I understand this correctly. The SIP RTP ports are dictated by whatever I set on the Asterisk server ? So there is no need to set the ports on the Asterisk server, and then set the ports on Jitsi to match ??

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-----Original Message-----
From: users-bounces@jitsi.org [mailto:users-bounces@jitsi.org] On Behalf Of Damian Minkov
Sent: Monday, November 03, 2014 8:20 AM
To: Jitsi Users
Subject: [PossibleSpam] Re: [jitsi-users] [PossibleSpam] Re: SIP RTP Port Range

Hi,

if you edit the file by hand, you need to stop Jitsi before that. The local media ports has nothing to do with the ports that asterisk uses, both are for local use only. The ports used in the call are negotiated and can be in different ranges without affecting the call.

Regards
damencho

On Mon, Nov 3, 2014 at 2:49 PM, Robert Seaton <bobs@itproscorp.net> wrote:

I did find out where to edit the configuration to add the settings. Thanks.

-----Original Message-----
From: users-bounces@jitsi.org [mailto:users-bounces@jitsi.org] On
Behalf Of Robert Seaton
Sent: Monday, November 03, 2014 7:10 AM
To: Jitsi Users
Subject: Re: [jitsi-users] [PossibleSpam] Re: SIP RTP Port Range

Yes, The SIP acounts in Asterisk are set to NAT. I'm testing with Jitsi, and 3CX SIP phone loaded on each computer. Once I get this config working I'll remove the 3CX phone, and just use Jitsi for both. The 3CX-to-3CX phone works fine. JitsiSIP-to-JitsiSIP does not. I also know that my RTP port range in Asterisk is set to 10000-20000. The settings you've provided? Do they need to be added to the (sip-communicator.properties) file manually? Sorry, I'm a Newbee.

Thanks for the lightning fast response!

-----Original Message-----
From: users-bounces@jitsi.org [mailto:users-bounces@jitsi.org] On
Behalf Of Damian Minkov
Sent: Monday, November 03, 2014 6:44 AM
To: Jitsi Users
Subject: [PossibleSpam] Re: [jitsi-users] SIP RTP Port Range

Hi,

by default the rtp port range is 5000-6000. But normally you do not need to know that.

A common problem with asterisk is that Jitsi is behind nat and this is not setup for that in asterisk. Do you have nat=yes for your sip account.

Port range can be controlled by configuration properties no UI setting currently for that. The properties with their default values are:
net.java.sip.communicator.service.media.MAX_PORT_NUMBER=6000
net.java.sip.communicator.service.media.MIN_PORT_NUMBER=5000

Regards
damencho

On Mon, Nov 3, 2014 at 1:31 PM, Robert Seaton <bobs@itproscorp.net> wrote:

I have two Windows clients. I'm running Jitsi version 2.5.5332. I
have two accounts setup on each client. Jabber connection to my
Openfire Server, and a SIP connection to my Asterisk server. The
Jabber account works fine. The SIP account does not(No Audio) How do
I change the RTP port range for my SIP account from within Jitsi?
Also what is the default port range for SIP account for Jitsi.

Thanks in advance

BobS

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