Know-how  | Topic: [Voice] | 4 MIN READ TIME
  

What is SIP Trunking?

What is Sip Trunking

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SIP stands for Session Initial Protocol. Widely recognised as the preferred method of communications transit in the modern era, SIP was designed by Jonathan Rosenberg in 1996 and has seen many iterations and enhancements. Today, SIP is the ready replacement for ISDN services and powers most cloud telephony solutions globally.

You should be familiar with ISDN trunks and PSTN lines. A common scenario for SIP use is replacing these legacy services to provide voice connectivity to your phone system.

Essentially, a SIP trunk is a virtual phone line. Calls are transmitted over the internet, without the need for copper cabling and infrastructure run into your building. This allows your business communications to benefit from cloud resiliency and functionality capabilities without fully adopting a cloud phone system. SIP trunks are the ideal middle ground when you need to upgrade your communications but have no compelling reason to replace your existing PBX.  

How does SIP work?

SIP is the protocol of VoIP communication that allows users to make and receive calls over the internet.

SIP works by sending messages from one SIP address to another. These messages are typically voice calls. However, SIP also powers messages in the form of video calling and instant messaging.

SIP is an over-the-internet exchange of information. A session is setup to initiate a transmission of these voice, video or instant messages. Once a SIP session is established, commonly referred to as a handshake, the data is sent, managed and ended by SIP. This sequence of events happens over milliseconds. This is how SIP calls are able to provide call quality better than that of a traditional phone line.

The parameters of SIP only include the call itself. Equipment or software (endpoints) are required at either side of the call (SIP session). These could be standard telephone to telephone interactions, a blend of desk phone to softphone or from dedicated SIP devices like conference phones out to the external PSTN network where non-SIP users are able to receive calls as standard.

SIP encoding

For SIP calls to be transferred over the internet, it must be encoded. This means speech must be transmitted as data. The audio signals found in speech are translated into codecs and passed from a SIP endpoint to its desired destination. For audio calls, SIP most frequently uses G.711 and G729 codecs. Other codecs are available for lower quality traffic and specialist high quality transmissions.

G7.11 is the standard codec used in SIP. For better quality of speech, this codec transmits the audio signals without compressing the voice data. Quality is not lost when using G7.11 as it utilises the bandwidth available to transfer speech as if in the same room as the caller.

G.729 is a reduced quality codec, where the voice data is compressed. G7.29 is typically seen in scenarios where bandwidth is restricted or unavailable for dedicated voice use. This is often the case in temporary deployments when clients have ADSL and FTTC connections rather than installing a dedicated internet connection.

For long term business voice, G.711 is always recommended. When moving a business to SIP, it’s important to demonstrate the quality, rather than cut corners on bandwidth. Typically, the savings achieved with SIP provide an immediate ROI – leaving budget for additional bandwidth if required.

SIP network readiness

As SIP only looks after the calling aspect, a crucial part of the SIP journey is your network. For SIP to operate at optimal performance, your network must be ready to let it do so.

It is important to segregate a portion of your internet connectivity for voice. This allows the application of to ensure the call quality is as best as it can be.

The amount of bandwidth you need to dedicate to SIP differs in every business. Working out how much is required is like planning ISDN trunks. Where you had 20 ISDN trunks, this meant you could have 10 people on the phone at any one time. If you still need 10 concurrent calls, 10 calls equate to 1MBps of bandwidth. 20 calls needs 2MBps, 30 calls needs 3MBps and so on. 

Lack of bandwidth, QoS or poor network configuration leads to degraded SIP call quality. Packet loss, jitter and latency have a negative on all parts of your network. As SIP calls power real-time conversations, the delay or quality loss is more obvious to the end user. Often, degraded network symptoms are displayed when calls drop mid call or audio is distorted and delayed.

Security within SIP trunks also relies on your network. SIP sessions themselves are secured by encryption of the SIP protocol for secure transmission.  Ensuring your underlying connectivity and network is secure end-to-end will allow SIP trunks to mirror this security.

Business Benefits

There are three immediate benefits of implementing SIP trunks.

SIP trunks are considerably cheaper than ISDN services. Both up front and monthly ongoing costs will significantly decrease. ISDN lines costs in excess of £200 per installation and at least £30 per pair, per month. SIP trunks are configured by your service provider on demand. This removes installation resource and charges. As SIP is an over-the-internet service, there is no rental from the likes of BT Openreach for your service provider to consider. This means the SIP trunk market has pushed the cost of SIP down to an affordable price for any business.

The quality of SIP calling is superior to that of traditional on-premises, infrastructure based calling. The ability to apply and upgrade codec, adjust bandwidth and apply QoS ensures voice quality is as clear as real-time, always.

ISDN services are due to be discontinued by BT Openreach by 2025. This is important for business for two major reasons.

Firstly, after 2025, ISDN services will no longer be supported. Therefore, when your phone lines have a fault, they will be irreparable. The likelihood of BT Openreach letting this scenario unfold is small, however. Whilst not formally communicated, it is likely that the entire network will be switched off to avoid unexpected issues.

Secondly, orders for new or additional ISDN circuits will likely cease in 2020. This means as your business grows and requires additional phones lines, you will be stuck with your current setup. Reacting to this realisation when you need to introduce new staff to your business is far from ideal.

The decision to move to SIP trunks ahead of time is one to seriously consider. For more information on SIP trunking and the future of business communications, check out our guide to getting started with SIP trunking.

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 Editor's notes: Last updated on March 20, 2019

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