VOIP Cloud-Hosted PBX

SIP is the IP protocol used to transport voice traffic over an IP network thus allowing users to make voice and video calls mostly for free over the Internet or a private network. This protocol is used by VoIP phones to communicate to the IP PBX for call registration and call setup/call teardown.

You may notice that many VoIP devices such as VoIP phones are also called SIP phones. The term VoIP and SIP do tend to be used interchangeably to describe the phones/handset.

RTP is the protocol that is used to transport the actual voice data over the network and works hand-in-hand with SIP that is used to register the phones and setup/tear down the calls

The SIP phone is the physical phone that sits on the users' desks. Rather than connecting to the legacy BT wall socket it instead connects to the LAN switch. When booted up, the SIP phone uses the SIP protocol to register its number with the cloud PBX and confirm it is online for calls.

There is a huge selection of SIP phones on the market today. At present we recommend the Cisco SIP phone range as they are extremely stable and offer excellent call quality and user overall experience.

A SIP client is available for locations that do not require a physical phone. This is SIP software that runs on a PC or Mac to simulate a phone allowing inbound/outbound calls and all the same additional features possible with a physical phone handset.

Most users still prefer a physical phone but for hot-desking or locations that are hardly used then a softphone client is an excellent choice.

You can simply plug your new VoIP phones into your LAN switch and away you go. Ideally, your LAN switches will have 2 features to get the best out of your VoIP service.

Using ‘managed’ switches rather than ‘un-managed’ switches can allow the LAN infrastructure to be configured to prioritise voice and video traffic using Class of Service (CoS). We can fully assist you with this aspect should you need it.

The PoE (Power over Ethernet) feature allows your VoIP phones to obtain power from the LAN network cable rather than needing an additional power supply. This is an excellent feature that we recommend to your current and future switches support.

Each of your sites should have a good connection to the Public Internet or a Private MPLS network (more on this later) to ensure enough bandwidth for your voice and video conferencing traffic.

Your network provider should configure Quality of Service (QoS) on the WAN router to prioritise the voice and video traffic as it exits the router. We perform this by default for all our customers.

The VoIP Cloud Provider hosts the VoIP equipment on their own network and is responsible for the management and support of the entire VoIP platform. If there are any issues you would report them to the provider rather than troubleshoot yourself. You are responsible only for simple admin tasks and provisioning orders for new services/phones via the providers portal.

Your SIP Phones or SIP clients register with the Cloud Provider’s VoIP PBX to obtain their number and the establishment of all calls is via the Cloud Providers platform.

Businesses that make mostly external voice and video calls rather than site-to-site should consider this model as it is quick to provision and has all the features you will need.

If your company has multiple sites that will communicate with each other for voice and video, then a cloud-hosted VoIP solution on a private MPLS network is highly recommended. All of your company's site-to-site calls and video conferencing data traverses a private high-speed MPLS network rather than using the Public Internet which allows for better overall voice and video performance, monitoring and enhanced SLAs.

Hunt Groups are useful for businesses that don't require a call centre solution or lots of functionality, hunt groups efficiently distribute calls received by a single number amongst a group of users. 

Hunt Group Features include:

• Call waiting
• Call forwarding
• Custom ring back
• Pre-alerting announcement

Time of day routing is included with optional voicemail add on if required. 

With all the core features of hunt groups, call centres go even further with announcements, on-hold music, queueing and flexible add ons to help businesses scale to their call demands.

The key difference between call centre queue levels are the number of calls that can queue at any one time:

Basic queue - 25 calls per queue
Enhanced queue - 50 calls per queue
Premium queue - 525 calls per queue

IVR (Interactive Voice Response) reduces the need for end-users to transfer calls by allowing the caller to choose the department or person they want to speak to.

Features include:

• Multi-tier choices
• Call forwarding selective
• Business hours rules
• After hours rules
• Holiday rules 

• These services provide a universal non-geographic dial-in number and the enterprise is charged for calls to these inbound numbers.
• If the call is then answered by an end-user, hunt group, call centre or IVR, there are no further charges.
• Alternatively, if the call is directed to an external number, that is charged as an additional call in addition to the initial inbound call.

• Support customer service, legal and training requirements with flexible call recording.
• Best suited for contact centres, legal, medical and any business with compliance needs.

CRM integration allows callers to be detected, looked up in a CRM system and details about the caller to be delivered to the user in a screen pop, or even automatically open a customer record. End-users install the Go Integrator plugin to benefit from this functionality. 

The Receptionist Web App is the perfect tool for receptionists or office managers who need to screen and manage calls.

Users can transfer a call efficiently just by clicking the destination in the contact list, there's no need to remember individual extensions. 

The Softphone add on is free for executive and collaboration service pack users, Communicator enables Unified Collaboration (UC) by bringing together powerful services:

• Instant messaging and group chat
• Presence
• Voice and video calling
• Multi-party voice and video conferencing
• Real-time desktop sharing

Communicator is available on PC, Mac, iOS and Android devices.

When you make the move to SASBOSS you will usually have a set of phone numbers you want to bring across. You can order new numbers from SASBOSS too.

DID products include:

• Individual non-consecutive numbers.
• A range of 10 or 100 consecutive numbers.
• 1300 and 1800 numbers.
• International toll free.
• Mobile numbers.

Mobile numbers can give you additional flexibility like:

• Using the number as a termination point (eg. for inbound calls answer on a queue, IVR or endpoint).
• Displaying the number when outbound calling using calling line id (CLID).