Company with many offices around the globe

[e23cx]

Hello Community,
I'm new here. I have just started testing 3cx for our company. I try to plan how it should be implemented in to our company which have offices in different countries. I plan to go with on premises solution. One 3cx installation in each office and then connect them all together via 3cx bridges.
Did anybody have this design before? can five 3cx systems be connected with each other? and can they all be managed from one GUI, or I will need to login to each 3cx system and manage it individually?
Is there an option to have only one 3cx system in one place in one country and put only SIP/PRI-Gateways in other offices (like beroNet)? beroNet is a PRI gateway, but do you know any SIP gateways?
Thank you in Advance!

 

[daktur]

Sorry but why would you 5 five systems bridged? You can do only one installation that will serve the entire company around the Flo. It makes more sense to me...
I understand bridging with other systems but not 3cx itself inside the same company.

 

[e23cx]

Hi Daktur,
I'm just not sure about the latency. For example: main office is in US, the 3CX will be installed in US, and remote office is in Asia. There's a VPN between offices with a latency about 250ms. In Asia we will buy a SIP trunk from a local provide. So this trunk will be connected to the 3CX system in US. When users in Asia office make a local calls, the voice traffic will go via US, right? that's why I thought installing a second 3CX in Asia won't route local calls via US.

 

[sip.bg]

You can do what ever you want, depends on your network connectivity, if you build IPsec VPN tunnels between offices and bandwidth is enough and connectivity is not breaking down, i.e. your internet providers are reliable.

Architecture should depend on factors like local lines, number of extensions, local survivability requirements etc. You may share some details, if you like (or in a private message).

If you have reliable network connectivity and local survivability is not crucial, you may use only one PBX, with SIP trunks from remote offices or VoIP gateways for analog / digital (ISDN) lines. This may require dedicated GRE tunnels between sites and several LAN cards in the PBX server (if IP autorization is used). If you run Debian you may use easily VLAN interfaces. Simplified scenario is to use 3CX SBCs, if possible. I would prefer not to use SBCs in a complex network architecture.

In all other cases you need to have several (five) PBXs and bridge them (3CX bridge). For enhanced features over bridges you may need Pro licenses.

 

[leejor]

If you do one central server, you have to ask, how reliable are the ISP's at each remote location. If a site losses connectivity, then all communication, even internal, is lost. Each location would also require an SBC, that may put a limitation on the number of extensions at each. If you planned no using a PSTN gateway at a location, keep in mind that is not supported, not to say you couldn't make it work, but you'd be on your own. Any outbound calls would still have to go back to the PBX because even though there is an SBC, I don't believe that a remote VoIP provider, at the same location can also be behind the SBC, so there is increased traffic, and a possible source for delayed voice.

 

[sip.bg]

Remote SIP trunks (supplied locally on a dedicated ethernet interface), VoIP gateways and large amount of remote extensions can be connected easily to central site (PBX) via VPN (GRE or other tunnels) -- different subnets can be routed without NAT.

This makes global network picture look like local site one for provisioning of extensions, trunks, gateways, etc.

Quality of network connectivity is crucially important for reliable operations.

 

[e23cx]

the ISP's are reliable on both sides (US and Asia). I plan to use a PSTN gateways on remote location in Asia like beroNet E1.
Or do you know any other voip gateway which could work with SIP?
I know Cisco CallManager has voice media gateways, you can use them on remote locations to handle Voice/RTP traffic without going to the server. But I don't know if 3CX has something similar.

 

[sip.bg]

the ISP's are reliable on both sides (US and Asia). I plan to use a PSTN gateways on remote location in Asia like beroNet E1.
Or do you know any other voip gateway which could work with SIP?
I know Cisco CallManager has voice media gateways, you can use them on remote locations to handle Voice/RTP traffic without going to the server. But I don't know if 3CX has something similar.

You can use any gateway you are comfortable with like Beronet and others. If this gateway working with 3CX locally, you can put it anywhere in the world using a VPN tunnel. Cisco, Avaya and others are not doing anything more. It is all about IP network connectivity. That simple.

Latency of 250 ms is critically big and may affect voice quality.

 

[e23cx]

You can use any gateway you are comfortable with like Beronet and others. If this gateway working with 3CX locally, you can put it anywhere in the world using a VPN tunnel. Cisco, Avaya and others are not doing anything more. It is all about IP network connectivity. That simple.

Latency of 250 ms is critically big and may affect voice quality.

Can you tell me, if I use a gateway(Beronet) on remote location, and a user on this remote location makes a call into PSTN via this gateway, how will the RTP go ? will it go via 3CX server or between Beronet and a user device on the same subnet?

 

[sip.bg]

You need to have VPN between offices (using appropriate routers). I can share experience separately.

If your local central site network is 192.168.0.0/24, the 3CX PBX has for example address 192.168.0.10
Remote sites can have networks like 192.168.1.0/24, other: 192.168.2.0/24, etc. The gateway (beronet) in remote office could have address like 192.168.1.10.
Configuring gateway task is exactly the same as if the PBX and gateway were in the same LAN (or a network with wider mask like 192.168.0.0/16).

Cisco Call Manager (express) is running on a Cisco router typically (larger implementations on a standalone server), voice media gateway is typically a card (gateway) into another Cisco router. Two routers are configured to build VPN between them, so Call Manager can communicate with Media gateway. Old implementations were using proprietary Cisco protocols like Skinny or H.323, not SIP.

The open system approach based on SIP standard and used in case of 3CX, Beronet (and other brands) has greater flexibility and more benefits and is more cost effective.

 

[sip.bg]

Regarding RTP flow, 3CX PhoneSystem forces SIP trunks and voice gateways RTP traffic to pass through the PBX. Only RTP traffic between two extensions can be peer-to-peer.

 

[sip.bg]

Make sure Beronet will work with ISDN in Asia or USA. It's not Euro-ISDN signalling, I'm not sure Beronet is supporting all international species of ISDN. Please check with Beronet.

 

[e23cx]

Regarding RTP flow, 3CX PhoneSystem forces SIP trunks and voice gateways RTP traffic to pass through the PBX. Only RTP traffic between two extensions can be peer-to-peer.

I read about 3CX media on this wiki page: https://www.3cx.com/blog/docs/media-server/
That's why I thought I could use some 3d party media gateway in Asia in order not to pass RTP traffic through 3CX in US. In case of "Proxy – Pass Through Mode" only SIP and RTP headers will go via 3CX, the voice(RTP) itself will go directly form SIP provider or beronet-Gateway to endpoints in Asia office directly.
Or am I wrong?

Can we talk offline, I would like to hear your experience with 3CX?
thanks

 

[sip.bg]

You may uncheck 'PBX delivers audio' into gateway settings, this should theoretically allow RTP to pass to extensions directly from gateway. Still needs to be checked and confirmed with Wireshark or similar tool. I will send you my emal in private message.