Jitter is a network traffic problem that, in VoIP communication, causes voice distortion, loss of call quality, and missing or jumbled words. Jitter is caused by voice data packets not arriving at their destination with a proper timing due to network interference and congestion. VoIP codecs can and do arrange out-of-order data packets. But with enough delay the codec will just not wait for the delayed packet.
Jitter is measured in milliseconds and generally packets delays beyond 50ms will cause noticeable problems with call quality. Jitter is caused by different network related factors which, in most cases, are controlled by your ISP and not by Telebroad. Still, our software and some of the hardware we support employ jitter buffering —a technology that temporarily collects voice packets in a storage area before playback, accounting for late arriving packets— to reduce or eliminate the phenomenon. We may also be able to suggest some tests and optimization on your network and firewall settings that may solve the problem. On your end, you should, at any case, try to avoid running your primary Telebroad's VoIP communication over WiFi.
A measurment of the average time, in milliseconds, it takes a (voice) data packet to travel over a network from the sender to the recipient. High latency (over 300 milliseconds) results in degradation of call quality due to noise, disturbances, and echo. Latency under 150 milliseconds is barely perceptible. Regardless VoIP calls are ideally operated with latency of under 50 milliseconds. Latency is caused by different network related factors which, in most cases, are controlled by your ISP and not by Telebroad.
Still, if you are experiencing noise and static we can advice you on some tests and adjustments on your network and firewall settings that may solve the problem. With the mobile version of the TeleConsole it is also possible to switch networks or change codec settings as explained here. You should also try to avoid running your primary Telebroad's VoIP communication over WiFi.
Local Number Portability refers to the option provided to a customer of a land, VoIP, or mobile telephone number to re-assigned to his number, as-is, to another carrier. LNP promotes healthy competition since it gives the customer the incentive to move to a better carrier without fearing of losing the telephone number recognized with his business. See this article and this one about advise and support provided by Telebroad when you port your numbers.
Media Access Control is a unique address given to and hardwired to every network adapter (the one inside a SIP phone in our case) when it is manufactured. Unlike IP addresses that can be assigned to a device and changed, MAC addresses are set. Theoretically there are no two identical MAC addresses in existence across the entire Internet. This is important so computing devices can pass information to each other. Even if duplicate MAC address exist, as long as they are on a different LANs, it won't be a problem.
Data transmitted over the Internet will use both IP addresses and MAC addresses to reach its destination. While the IP address defines its final destination, it may pass through numerous MAC addresses of various computing and network devices on its way to that destination. Usually a sticker on on the bottom or lower back of the your SIP or IP phone will show the MAC address, otherwise it can be checked in the phone's setting directly or via its web user interface. A MAC address contains the numbers 0-9 and the letters A-F. This helps avoid confusion since 0's in any MAC address are clearly the number zero and not the letter O.
MMS stands for Multimedia Message Service. It is an enhancement of the SMS protocol that allows users to send more than just textual information to and from mobile phones or VoIP platforms that supports it. MMS is most commonly used for sending pictures, but also supports video and audio recordings. Another advantage is that text messages can be longer than 160 characters.
Outbound Caller ID
For internal calls between users, Telebroad's PBXellent and other PBX phone systems use telephone numbers that are shorter than the ten-digits national format. However, for outbound calls with external numbers a full ten-digits Outbound Caller ID is assigned to users. This caller ID is different than a user's DID number and can be shared among the company's users. As such, it usually identifies the company rather than an individual user. For call back either the DID number will have to be provided to external callers or some routing conditions need to be set up by the phone system administrator to associate the Outbound Caller ID with a specific user.
A network connection method where data is divided into packets (segments) that are transmitted independently through the network. Each packet always contain the destination information and it's position in the data sequence. This means that, unlike Circuit Switching, each packet can take a completely different route through the network to reach from the source to the destination, where it is put into its correct position in the sequence. Packet Switching allows for greater network capacity (lines can be shared between packets from different sources) at the expense of lesser reliability and quality when lines get overloaded and packets are too late to arrive.
Peer (Dial Peer)
A peer or a dial peer on a telephone network is any device, software application, or phone system function that is able to either make, receive or transfer calls. It is a general term that can refer to both your desk SIP phone, your cloud voice-mailbox, or the queue that transfered the call to you. Peer is also known as addressable call endpoint. The "endpoint" part may be a bit misleading because some peers act as "in-between" points during a flow of a call. Peers are not limited to VoIP communication. Analog equipment connected to your network also contains peers. Analog peers are known as POTS peers.
Power over Ethernet (POE) is a popular technology that allow SIP telephones and door phones (or other networked communication devices) to receive electrical power over their network cables instead of a traditional power cord. This provides for more placement flexibility since the telephones don't need to be near a power socket. To see which SIP telephone sold by Telebroad support POE please consult our buying guides for telephones and door phones.
Public-Safety Answering Point (or "Public-Safety Access Point") is a call center that accepts and handles emergency 911 calls from mobile, landline, and VoIP subscribers. The PSAP is responsible for dispatching emergency first responders to location of the subscriber, providing them with an address obtained from either the ALI database or by VoIP or mobile network geo location technology.
Plain Old Telephone Service is a term used for traditional analog voice telephone service that is transmitted over copper lines. POTS was the standard in telephony communication since the dawn of the industry until new technologies emerged in the 80's. While lacking the advantages of those technologies, POTS offers remarkable reliability and the ability to function even in power outage because the current for its operation is supplied by the telephone company. Existing POTS line in your office can be integrated into a VoIP network using an ATA device.
Public Switched Telephone Network. The traditional telephone network which uses underground copper wiring to carry analog or digital T1 signals. It uses the Circuit Switching connection method and comes with dedicated lines for each device on the network, commonly referred to as land lines. It is more expensive to use, especially on long distance or international calls. A business that uses the PSTN will have to pay for additional physical lines when expanding beyond the capacity of its PBX. The advantages of PSTN is that it offers more reliability, functionality during power outages, and 911 location tracing.
P2P stands for or person-to-person messaging. It is a type of two-way text messaging conducted between two people, as opposed to A2P messaging. The messaging takes place on mobile phones or between SMS enabled VoIP platforms (such as Telebroad's Business SMS).
QoS and QoE
Quality of Service and Quality of Experience are two related terms that measures the data delivery efficiency of a network. QoS is examined from the perspective of the service provider (Telebroad or your ISP) while QoE is examined from the perspective of the user (you or your company). QoS quantify the network performance with measurable metrics such as packet loss or corruption, latency, jitter, congestion, etc.
QoE, on the other hand, uses more subjective measurements deriving from a user's expectations, such as voice quality, noise levels, responsiveness etc. Since voice quality and noise levels depend on network efficiency (less latency, jitter etc.) a provider's ability to improve QoS will automatically improve QoE (again within a user's expectations). Is is possible for QoE improvment by adjusting the network to prioritize certain packets. In VoIP in particular this would entail prioritizing packets tha carry voice information.
API stands for Application Programming Interface. It is a method of communication, using specific commands and format, between software components on a computer or between a client and server over the Internet. REST stands for REpresentational State Transfer. It is a kind of API that is specifically designed as a web service, allowing a program to communicate directly to a web page, without needing to use a browser or having a user interact with the page.
APIs requests lack the visual aspect of web sites and only return pure data from the server or perform a specific service on it, acknowledged by nothing more than a textual confirmation. Regardless, REST APIs are a very powerful tool and can obtain, update, or create data in a very efficient manner. They also allow integration between completely different software or online services platforms.
Telebroad offers users and developers a comprehensive set of VoIP and communication REST APIs based on our TeleConsole software. See this article for summary of functions and descriptions, this one for how to use API requests, or this entire section for complete documentation.