[OpenSIPS-Users] register phone in same local network as opensips

Alexander Jankowsky E75A4669 at exemail.com.au
Sun Apr 23 13:01:16 EDT 2017


 

Thankyou for your reply. It got me thinking for a few days and a lot of testing.

It looks very much like the router at the remote phone has some form of Sip ALG.

Finding with that brand in general, having a reputation for locked enabled Sip ALG.

The change is very subtle, easily glossed over and so it took some understanding.

I have all the ports sorted out correctly now, so all that is all working properly.

 

On Wireshark for inbound calls the rtp arrives correctly and actually has audio on it

but the phone itself does not reproduce the audio at all and it is totally silent and

there is no local outgoing rtp stream at all. Outbound calls now all work correctly.

I even swapped phones over and once again had exactly the same problem.

 

There is no problem at all when we use direct IP to IP calling through this router.

So, this is somewhat of a larger problem, I am now looking at implementing TLS.

 

Alex

 

 

From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Newlin, Ben
Sent: Friday, 14 April 2017 10:04 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] register phone in same local network as opensips

 

It doesn’t sound like it has anything to do with the registration. It sounds like your router has some sort of SIP Helper application that is trying to assist by re-writing the ports in the INVITE. Many modern routers come with this functionality enabled by default, even though in my experience it does nothing but break SIP communications.

 

Take a look at your router documentation for any mention of SIP functionality and disable it.

 

Ben Newlin 

 

 

From: Users < <mailto:users-bounces at lists.opensips.org> users-bounces at lists.opensips.org> on behalf of Alexander Jankowsky < <mailto:E75A4669 at exemail.com.au> E75A4669 at exemail.com.au>
Reply-To: OpenSIPS users mailling list < <mailto:users at lists.opensips.org> users at lists.opensips.org>
Date: Friday, April 14, 2017 at 8:08 AM
To: " <mailto:users at lists.opensips.org> users at lists.opensips.org" < <mailto:users at lists.opensips.org> users at lists.opensips.org>
Subject: [OpenSIPS-Users] register phone in same local network as opensips

 

 

 

Hello,

 

I have opensips 2.3 beta, along with a local phone both running inside and behind the same router.

I have one port forwarded for opensips to listen on and a port range forwarded for the local phone.

There is a remote phone in another domain behind another router, also with a port range forwarded.

I am using stun for both phones and this resolves the correct IP domains for each phone.

With stun implemented and saying it is full cone on both phones. The phones can now ring each other.

 

The local phone can call the remote phone and there is two way audio.

When the remote phone calls the local phone, there is neither way audio.

Invites from the remote phone always appear with the correct expected provisioned sip and rtp ports.

 

I would expect that the local router is changing the local phones sip contact port when it registers.

When I look at a sipgrep capture of an outgoing invite both the sip and the rtp ports are changed.

I am not all that sure where in the process or even why the rtp port for the invite has been changed.

Inbound calls then of course end up sending and returning rtp through non forwarded port ranges.

 

What I would like to understand is how to make an inclusion, when any local phones register,

that will allow the outgoing contact details to show the phones actual provisioned sip ports.

With that correct, in the outgoing invite, the rtp streams would then normally be within the

expected range of ports opened and forwarded to the phones and that would solve the audio.

 

I am looking for working examples, but I have not turned up enough specific information about just this.

Knowing better where and how to start and the names of what I am looking for would be most helpful.

 

Thankyou

Alex

 

 

 

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