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SIP audio not reaching destination.

RogerWoods
Newbie
Posts: 4
Registered: ‎01-04-2023

SIP audio not reaching destination.

SIP is short for 'Session Initiation Protocol' and is widely used to send/receive VoIP/AoIP voice/audio call across the internet.  If you have VoIP phone on your desk a home/work SIP is what makes it work.

SIP is also widely used in broadcasting, using software/hardware audio codecs to setup high quality peer-to-peer audio links between radio studios/stations, reporters, guests, big events, etc., etc. And, this is the root of my question as I'm hoping someone in the tech department might have some insight that could help me.

So, the situation is this; Oh, before I start I'm a PlusNet Customer of over 10 years. 

As a broadcaster who hosts a daily live programme from my home studio i have some interesting kit to play with.  Plusnet enables me to get an audio stream over the Internet to the radio station network centre for rebroadcast on DAB across Suffolk, UK.

I'm also a bit of a tech nerd when it comes to broadcast audio and the station is presently experimenting with peer-to-peer audio connections using some hardware audio codecs that are SIP enabled.  My present scenario is this; I have a hardware audio codec box connected to my home router (TP-Link VR2800), with port forwarding enabled to route any external requests to my static public IP address on ports TCP-5060 and UDP-9150 forwarded to the internal IP address of the codec (212.XXX.XXX.211 > 172.16.1.XX).  This works well as users of the same codec are able to call my box and establish a real time low latency high quality bi-directional audio stream from anywhere in the world. I have received calls from the USA, Europe, New Zealand and, from another box I have setup on a neighbours broadband from a different IPS.  It works using 4G/5G mobile data too.

The problem I'm having is while I can receive calls from all over the internet, I cannot make calls to other users IP addresses from my location.  I can, of course, make calls on my internal  network, but not on the public network.   

Does anyone have any insight on what may be causing the problem of not being able make SIP calls out? This is on both my FTTC broadband and 4G/5G connections. 

 

7 REPLIES 7
MisterW
Superuser
Superuser
Posts: 14,698
Thanks: 5,493
Fixes: 393
Registered: ‎30-07-2007

Re: SIP audio not reaching destination.

@RogerWoods have you got the sip alg enabled on the vr2800 ?

If so, try disabling it . You can find this setting in Advanced>NAT settings>alg , I think.

Superusers are not staff, but they do have a direct line of communication into the business in order to raise issues, concerns and feedback from the community.

RogerWoods
Newbie
Posts: 4
Registered: ‎01-04-2023

Re: SIP audio not reaching destination.

It's never been enabled;  SIP AGL is well known for strippping SIP packets in unexpected ways, corrupting them and making them unreadable.

 

MisterW
Superuser
Superuser
Posts: 14,698
Thanks: 5,493
Fixes: 393
Registered: ‎30-07-2007

Re: SIP audio not reaching destination.

Which is exactly why I asked the question.

I've used voip(sip) at home for about 5 yrs now, and also manage a small office  pbx so have a fair knowledge of how sip works.

Do you have any more info on the codec box ?

Superusers are not staff, but they do have a direct line of communication into the business in order to raise issues, concerns and feedback from the community.

RogerWoods
Newbie
Posts: 4
Registered: ‎01-04-2023

Re: SIP audio not reaching destination.


Hi Mr W.

I too have been using SIP audio for broadcast in various guises for over 12yrs and have very little issues before now.

The codecs in question are Telos Zephyr xStream, they were primarily ISDN boxes but have Ethernet sockets to enable SIP/RTP audio connections over LAN/WAN & internet.

They’re dated devises that when new were over £4000, online auction sites sell these for as little as £50. It they can have a second lease of life as IP codecs which most users/owners don’t realise.

I’ve have used SIP enabled product from Comrex, AETA, AEQ, InQodec and ipDTL; in fact I represent the later too as an ambassador for their products and services. All use a STUN server to enable calls. The Telos boxes do not.

It’s strange that I can receive calls but not make calls using destinations static or dynamic IP addresses.

It’s no great shakes as folk can call me, it I would be good to find out what the issue could be. I suspect it could be ISPs blocking services.

Any assistance or insight welcomed.

R.
MisterW
Superuser
Superuser
Posts: 14,698
Thanks: 5,493
Fixes: 393
Registered: ‎30-07-2007

Re: SIP audio not reaching destination.

All use a STUN server to enable calls. The Telos boxes do not.

my first thought when you posted that, was thats not going to work very well, especially in direct sip calling. It needs to get the wan ip in some way !

Seems like according to this, you have to configure it manually which is ok if you have a static ip ( which you do )

https://www.procommvoices.com/setting-up-a-telos-xstream-for-ip-connections/

Have you configured the wan ip ?

Superusers are not staff, but they do have a direct line of communication into the business in order to raise issues, concerns and feedback from the community.

RogerWoods
Newbie
Posts: 4
Registered: ‎01-04-2023

Re: SIP audio not reaching destination.


I do indeed have the WAN address Included in the codec settings.

Although you can also level it blank if you on dynamic addressing.

I have a second box on a 4G/5G router that without the WAN address and that connects perfectly to my box. But again I can’t call it on the number the codec tells might the call of from. It I put that down to the mobile network being different.
MisterW
Superuser
Superuser
Posts: 14,698
Thanks: 5,493
Fixes: 393
Registered: ‎30-07-2007

Re: SIP audio not reaching destination.

Most mobile networks use CGNAT so calling into a device on a mobile network is likely not to work.

TBH  the problems with SIP are usually for incoming calls rather than outgoing. Short of trying to get a packet capture to see exactly what is being sent when the codec tries to make a call, I'm running out of ideas...

Superusers are not staff, but they do have a direct line of communication into the business in order to raise issues, concerns and feedback from the community.