Navigate the world of telco like a pro
Telco is a tough world for a newcomer. While each and every one of you has at least a basic understanding of telco’s key concepts, (SMS is not included in this reference), speaking “fluent telecommunication” is a horse of a different color.
Indeed, obscure acronyms are everywhere. Googling them, you end up with a reference to a bunch of other acronyms or cryptic words, like, “Dual-Tone Multi-Frequency” (DTMF): you’re in for quite a Wikipedia ride if you want to fully grasp their meaning.
Acronym references do of course exist, but they are either too bare-bones to really be useful, (“DTMF: Dual-Tone Multi-Frequency”), or else too exhaustive to the point of being unusable. We decided to play goldilocks and create an acronym reference dictionary that gives just enough. We aimed to provide a quick definition of the concepts behind each acronym, and whenever relevant, include additional resources if you’re interested in further exploring a given topic.
We hope our acronym reference will provide newcomers in the industry as well as veterans looking for a quick answer a helpful guide to deciphering each acronym they may encounter – and spare them hours perusing Wikipedia.
API (Application Programming Interface)
“UIs (User Interfaces) are how people use computers. APIs are how computers use computers.”
An interface allows interacting with an object without knowing anything about it (such as the remote which gives control over the TV). An API is a communication standard for programs to talk to each other. It’s designed to structure and formalize communications so one program’s functions can be integrated into another. At CALLR, we offer an API to help brands create smart interactions through voice and SMS.
CDR (Call Detail Record)
CDRs are the data produced during phone calls. It contains all the call attributes such as the call time, its duration, the hang up time and cause and much more. CDR have two main interests: their analysis can provide insightful data, especially at scale and they are essential to diagnosing and debugging a faulty telephone system. On our Knowledge Base, we documented how to access and interpret CDRs generated using CALLR.
CLI (Caller Line Identification) / CID / CLID / CLIR / Caller ID
CLIs are a method to identify the number originating a phone call. The CLI can be a phone number of any type or hidden which results in a phone call originating from an unknown number.
For SMS, Sender IDs are used to identify the sender: they can be alphanumeric (example: “Your Bank”) to allow for the easy recognition of the sender by the recipient. If the sender wants to allow replies, the Sender ID must be numeric: any types of phone numbers, depending on the SMS nature and the country’s legislation.
DID (Direct Inward Dialing Number)
Originally, DIDs allowed companies to attribute a range of numbers to each person or workstation without requiring a physical line into the PBX for each possible connection. Nowadays, it usually designates phone numbers which can be bought or port in.
DLR (Delivery Receipt/Report)
DLR is a piece of information sent back by the mobile network to the original sender of an SMS. It contains the delivery status of the message making it invaluable to check whether the sent SMS has reached the destination handset. SMS Delivery timeline is also included: date created, sent, received and last update (useful if the SMS is put on hold because the receiver mobile is off, for instance). Webhooks allow for the creation of event-based notifications, such as the reception of the DLR of an SMS sent.
DTMF (Dual-Tone Multi-Frequency)
DTMF is the system used to communicate keys pressed on telephones. Each key is identified by a specific combination of two tones inherited from its row and column. Essentially, DTMF is the new standard that put the good old rotary dialer to rest. It can be used to create IVR scenarios to handle incoming calls: each key is bound to a different input. See more below.
E.164 (International Numbering Plan)
E.164 is the international numbering plan. Using a prefix system to reference the country of the called numbers, it allows any numbers to be called from any location. A number, following the E.164 format is formatted this way: + [1-3 digits (country code)] [max 15 digits (subscriber number)].
IVR (Interactive Voice Response) / VRU (Voice Response Unit)
IVR is a technology that allows humans and computers to interact over the phone using DTMF and voice inputs. IVR is used for mobile purchases, information, banking, and services. It’s also used to prequalify calls in order to route them to the best-suited agent. Both inbound and outbound calls can use IVR, although inbound IVR (IVR on incoming calls) is much more frequent.
LVN (Long Virtual Number)
A long virtual number (ex: +44 9865 80044) is a reception mechanism used by businesses to be able to receive SMS messages and voice calls. While LVNs are harder to remember than shortcodes they have some advantages, such as their international availability (shortcodes are country-bound) or their uniqueness (shortcodes are sometimes shared).
PSTN (Publicly Switched Telephone Network)
PSTN is used to designate the aggregate telephone networks operated by local, regional and national providers, providing the required infrastructure for telecommunication. It uses standards to allow different networks in diverse countries to interconnect seamlessly such as E.164.
MO (Mobile Originated) and MT (Mobile Terminated) SMS
MO and MT messages are frequently used terms by the industry professionals. While the acronym can be puzzling, their meaning is pretty straightforward. Mobile Originated messages refers to SMS sent from a customer’s phone (originated from an actual, real mobile phone) into your SMS handling system. Logically, Mobile Terminated messages designate messages sent from your system to a customer’s or prospect’s mobile (terminated on their mobile phone).
QoS (Quality of Service)
The Quality of Service refers to the overall performance of a telephony network as seen by the end users. QoS is evaluated using all aspects of a connection: response time, signal-to-noise ratio, echo, crosstalk, interrupts… QoS is critical for VoIP, especially in SIP.
VoIP (Voice over Internet Protocol)
VoIP is a methodology and a group of technologies used to deliver voice and multimedia communications over the Internet Protocol instead of the PSTN (see above). It relies on numerous protocols described in this article including SIP – session initiation, description and parameter negotiation, and RTP – media transfer. It’s used by voice and conference calling apps such as Skype or Messenger.
ACD (Average Call Duration) and ASR (Answer-Seizure Ratio)
As its name suggests, ACD is the average length of calls made in a given context. On the other hand, ASR measures call quality. It’s the percentage of answered phone calls with respect to the total call volume. We wrote a dedicated article explaining why good ASR and ACD scores matter.
IAX (Inter-Astrisk eXchange)
IAX is a communication protocol native to the Asterisk PBX. It’s a VoIP implementation particularly adapted for installations with limited available bandwidth. Indeed, IAX uses only one stream for both signaling and media payloads, making it much less resource-hungry. Moreover, IAX uses the Password Authentication Protocol, it doesn’t rely on IP for authentification. It supports NAT and firewalls.
NAT (Network Address Translation)
NAT is a method of remapping one IP address space into another by modifying the network address information in the IP packet header. Essentially, it’s a technique used to reroute traffic in IP networks without having to change the address of the re-routed host. With the limited amount of IPv4 addresses available, it became a useful method to face the shortage (read the Wikipedia article).
Besides, NAT is often used to refers to IP masquerading. It consists of hiding (private) IP addresses behind another public IP address. The public address is then displayed as the source by packets emitted using the private addresses. While there are security benefits, it has consequences for the quality of the Internet connectivity and makes the troubleshooting of the installation more difficult: NAT is a most of the time troublesome to conciliate with a VoIP setup.
PBX (Private Branch Exchange)
PBX are the interfaces between the corporate telephone network and the public telephone network. The PBX connects telephones within the business (short numbers on the local phone to call colleagues) and outside of it, to the PSTN allowing its users to call any number they may need.
RTP / SRTP ((Secure) Realtime Transfer Protocol)
RTP is with SIP one of the technical foundations of Voice over IP. It’s a network protocol made for delivering audio and video over an IP network. Once a session is initiated with SIP, RTP is used to handle the transfer of the audio/video stream from both sides. More details on RTP.
RTCP (RTP Control Protocol)
RTCP is a sister protocol of RTP used to monitor transmission statistics and quality of service. It’s also involved in the synchronization of multiple streams: audio and video or conference calls. Its primary function is to provide information on the quality of service (QoS) by periodically sending feedback on the media distribution.
SDP (Session Description Protocol)
SDP is a format for describing streaming media initialization parameters. Its main purpose is to help session announcement and invitation as well as parameter negotiation. While initiating a voice or video call over IP, both parties need to be able to understand each other, i.e, communicate using media type, format and codecs understandable by both parties: SDP is the protocol used to standardize media description and parameter negotiation. More details on SDP.
SIP (Session Initiation Protocol)
SIP is a signaling protocol designed to set up a communication over the Internet Protocol (IP). It can be used to initialize a voice, video or text communication but voice is the most common. It is here to ensure two main requirements necessary for a conversation: both callers know each others’ location over the network (IP address) and both can understand each other (they use the same codec). SIP works in conjunction with other protocols such as the (S)RTP ((Secure) Real-time Transport Protocol) which is used to deliver the audio stream over the IP network. More details on SIP.
While the acronym refers to the initiation protocol, “SIP” is used in the industry to refers to voice conversation over the internet in general – the SIP trunk, registrar SIP and IAX.
Looking for a SIP Trunking provider?Check out CALLR
Looking for more resources to understand the world of telecom better? Check out the Learning Hub section of our Knowledge Base: it addresses key telecom concepts such as numbers, DLRs, character encoding or codecs.