VoIP In General
Voice over IP, or VoIP for short, is a method and group of technologies for delivering voice and video communications over IP (Internet Protocol) networks, such as the internet or a company's local network.
Session Initiation Protocol, or SIP, is a communications protocol used for signaling and controlling voice and video communications. It is the most widely-supported VoIP protocol. It does not 'carry' the packets of data that contain the digitized voice or video on a call but it handles signaling (ringing, hangup, etc.) and negotiates the voice or video codec that will be used by each endpoint.
A trunk, or more usually referred to as a 'SIP trunk', is a digital pathway used for making VoIP calls to or from the PSTN or another PBX. Any call that is not made 'on net', or purely via IP networking, will need to interface with the PSTN and will need to use a trunk. All ITSP's provide a certain number of trunks to their customers' PBX systems, which determine the maximum number of concurrent calls to the PSTN that can be supported. In older PBX technologies, a company might have a PRI with 23 'channels' for carring voice traffic, which would mean a maximum of 23 concurrent calls to the PSTN. Since these PSTN pathways are now based on IP networks, (i.e., SIP trunking), this means that call capacity is much more flexible and can be increased as needed on the fly, provided that other requisite services (bandwidth, PBX hardware specifications, etc.) are able to handle the increase. For companies that need multiple PBX systems, SIP trunks can also provide inter-PBX calling without traversing the PSTN and thus not incurring extra costs.
Latency refers to a delay in the delivery of data packets between two networks. In the VoIP world, latency is a core metric to consider when deploying a VoIP PBX. All audio and video that is IP based is sent in data packets across an IP network, like the internet, usually using RTP (Real-time Transport Protocol). RTP 'streams' the packets to the endpoint without a mechanism for re-requesting a packet or packets should one be lost in transit. If there is latency between the two networks, these packets can arrive out of order or be dropped altogether, resulting in poor call quality or even dropped calls. Latency is usually measured in milliseconds and Virtualtone can provide a speedtest that will measure network latency for existing customers and for prequalification of potential customers. One other note: Latency is indpendent of bandwidth. Bandwidth refers the capacity of a circuit to carry parallel data between networks. Bandwidth is measured in bits per second (pbs), up to Gigabits per second (Gpbs), and having a circuit with a large bandwidth does not guarantee a good customer experience with VoIP. Please contact Virtualtone about our speedtest or to get more information.
Working with Virtualtone
If you have a current, manufacturer-supported phone that Virtualtone supports here then you can pay a one-time provisioning charge and Virtualtone will provide firmware (but not a warranty) for the phone. If the phone is from a different manufacturer, Virtualtone will provide the credentials to allow the phone to 'register' with the PBX, but Virtualtone will provide only limited support for the device for troubleshooting of bad call quality or other issues that might arise. If the phone is End of Life or no longer supported by the manufacturer (Meaning no continued support or updates of the firmware), Virtualtone will not allow the phone to connect to your Virtualtone PBX because of the inherent security risks of using unpatched phones. All phones purchased through Virtualtone are the current product lines for the respective manufacturers and all manufacturer warranties are handled by Virtualtone.
Virtualtone strongly recommends purchasing our own firewall, as it is tuned perfectly to the needs of VoIP and allows for remote troubleshooting of issues should they arise. Customers can configure their own routers if they choose using the guidelines here. Business-class routing and switching equipment should be used for optimal performance.
If you are planning any network change(s) please contact Virtualtone to let us know. We can discuss your change(s) and make plans for downtime and monitoring to ensure your phones continue to work.
Virtualtone addresses this in multiple ways. For those customers that have multiple ISP's, our Virtualtone router can be configured to fail over traffic to your secondary ISP automatically when the primary circuit goes down. For those customers that do not have redundant ISP's, we can configure a Call Flow Control that will send all inbound calls to another destination, whether that is a voicemail box, a cell phone, or an answering service. We can also, in lieu of a global Call Flow Control, configure individual extensions to use the Follow Me feature to ring their cell phones. The Follow Me feature is also useful for people that are not always around their desk phone.
No, each device or user needs a unique extension number. If you need to ring multiple people for incoming calls look into Ring Groups or Queues.
Ring Groups and Queues are methods of answering high volume incoming calls. Ring Groups allow you to ring certain static extensions while Queues allow for dynamic members to be added or removed to the pool of extensions as needed. Queues also allow for better metrics for monitoring your group and more even distribution of calls based on your preferences.
Device & User mode is method of setting up your Virtualtone PBX where users are not intristically associated with a particular device or phone. This mode is useful if you use 'hot desking', meaning you have people that share the same phone, or if you have mobile users that have multiple devices (e.g. a desk phone, a SIP phone app on their cell phone, and a softphone on their laptop), though only one device can be used for that user at any one time.