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I want to run a PBX behind a firewall behind a DOCSIS modem. Our ISP does not want to deploy a TA / gateway at our site. Our ISP only uses the modem for analog telephony and nothing else. QoS for voice on DOCSIS lines is achieved by having two MACs for each modem. One address for access, one address for voice. Because of that it seems to me that an ISP cannot deliver QoS for voice.

However, we want to have a PBX. There are cable companies which provide my desired service. I suspect the reason for not providing a SIP trunk is that there is no direct routing to the customer's site. Is it possible to run a SIP trunk under these circumstances?

WAN ---> DOCSIS 3 ---> modem as bridge ---> service IP by DHCP ---> customer's own device. Subnet routed via service IP.

The customer's router is connected to a DOCSIS modem in bridge mode. By requesting an IP via DHCP on the WAN interface, a varying service IP is assigned which is the last hop to the customer's booked subnet. The subnet is statically routed via a service address. Therefore, no public IP can be assigned to a WAN interface.

Usually, a VoIP solution independent from the ISP requires the reservation of an IP within the customer's network at the SBC for billing and QoS - as it is common practice via DSL lines. Considering the indirect routing and DOCSIS in between, is a SIP trunk possible?

  • Parties involved: ISP running the WAN and the SBC. Customer running the router and the PBX.
  • 200.0.8.0/28 is a public network.

Topology

  • Router's LAN interface and the PBX share the same network permanently routed routed via the service IP by the ISP. PBX's address is supposed to be listed by ISP's SBC to allow VoIP.
  • Router requests an address via DHCP and is assigned the service address which basically is the same one but may change due to technicalities, meaning it is not a static public IP in the narrow sense of the word.
  • Modem is invisible to the Internet and inaccessible to Router.
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    This question would be easier to answer if we had a diagram which includes the aforementioned components as well as who owns each of them – Mike Pennington Apr 13 '14 at 14:15
  • Edit: Added a picture of topology plus commentary in section below. – user2964971 Apr 13 '14 at 15:39
  • I could be missing something here, but as long as A) The PBX has been given a public IP from the customer's /28 (can I assume this is a public subnet?). And B) The static route for this subnet is automatically updated to point to the new next-hop IP when it changes, I can't see any reason why this wouldn't work. – Geekman Apr 13 '14 at 22:26
  • The only issue you might have is how restrictive your upstream SIP Trunk provider is. Some will only connect over a cross-connect into their network, others will allow connectivity over the internet. That's something only the provider can say. Ultimately, you could connect a SIP trunk even with a totally dynamic IP because the PBX would notify the provider what it's external IP is on each registration (of course this is not a recommended scenario, but I'm just saying it's possible, so I can't see you having any issues). – Geekman Apr 13 '14 at 22:31
  • Edit: A) Answered geekman's question about 200.0.8.0/28 being public. B) Clarified that SBC is run by ISP. – user2964971 Apr 14 '14 at 20:31
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To answer your question: it is possible to run a SIP trunk under your circumstances because your circumstances are no different than those of any other cable subscriber.

DOCSIS is a Layer 1 & 2 protocol, while SIP trunks essentially operate over IP & UDP which are Layer 3 protocols. I don't think it helps to mix them.

With most SIP trunking providers, when the PBX on your LAN registers with the SBC, the SBC uses NAT punching techniques to find the right private IP, if there is a NAT to cross.

Your deployment concerns are going to be:

  • Making sure your router doesn't interfere with SIP traffic (disable SIP ALGs - most implementations break SIP)
  • Hoping that your ISP doesn't traffic shape SIP
  • Checking that your connection to the SBC is solid enough (<30ms RTT, <1% packet loss, <10ms jitter)

Since you are concerned about QOS - this is something that is simply not possible with packets over the internet. Many SIP trunking providers will sell a managed service product that comes with private data T1 connectivity to their backbone - so essentially they're taking & owning the QOS problem.

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  • It does work because all I should have to do, is to name the IP of the PBX used for registration. That address will be authorized. – user2964971 Aug 10 '17 at 10:46

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