With voice over IP (VoIP) being a hot topic in the unified communications market, many businesses may be looking to migrate from a PSTN telecommunication provider to a SIP trunking provider. In some cases, a VoIP migration involves replacing the whole PBX system with a newer generation VoIP PBX, while some businesses might instead choose to keep costs down by interfacing their PBX with a SIP provider.
During this transition, IT departments are often overwhelmed with the issues that can arise. Here we will provide some key points that aim to make a VoIP migration as painless as possible.
PBXs that do not support SIP trunking will require a voice gateway to connect to their SIP provider. The voice gateway is placed between the PBX and SIP provider, connecting with the PBX on one end (usually using T1/E1 ISDN lines) and the SIP provider on the other end via IP:
PBXs that do not support SIP trunking will require a voice gateway to connect to their SIP trunking provider.FIREWALL.CX
It is important to ensure the voice gateway supports an equal amount of concurrent calls on both legs: SIP provider–voice gateway and voice gateway–PBX. The second leg, voice gateway to PBX, is where we usually find T1/E1 interfaces that provide a capacity of 24/32 concurrent calls respectively. If additional call capacity is required, multiple T1/E1 lines are installed, assuming the PBX can handle this capacity.
Selecting the right SIP service provider is one of the most important steps. The SIP provider must provide a stable service without interruptions, but must also own the physical delivery medium to the company’s premises, also known as the last mile, in order to guarantee quality of service (QoS).
Redundancy is imperative, especially for SIP trunks with large session counts. For session counts exceeding 20 (equivalent of a T1 connection — 24 channels) or 30 (equivalent of an E1 connection — 32 channels), it is recommended to have one SIP trunk on standby.
If the SIP session count exceeds 500 (equivalent of a T3 connection — 672 channels), then it is recommended to have separate SIP trunk entry points into the network. In some cases, the redundant SIP entry point can also be at different geographical areas. If the SIP trunk is down at one site, the SIP provider can temporarily route calls from that site through the SIP trunk of a nearby site.
If migrating to VoIP involves replacing a whole telephony infrastructure, extra caution is needed to guarantee a smooth switch to the new system.
Never rush into a production deployment. Set up the new VoIP infrastructure parallel to the existing PSTN/analog system. This strategy will give the company time to test the new system, decide on call flows and make adjustments to user rights and other VoIP services.
Also heed the network infrastructure. Voice packets must be in their own separate VLAN, usually named the voice VLAN. Network switches and routers must have QoS enabled and configured to prioritize the processing and delivery of voice packets.
Power over Ethernet (PoE) is another important feature to consider in a VoIP migration. If the local network switches support PoE or there are plans to replace them with PoE-capable switches, note the power requirements of each IP phone and ensure the switchport is able to meet them. In addition, the total power requirement per switch will depend on the number of IP phones connected to it, and this should not exceed the PoE budget the switch is able to deliver.
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