In voice telecommunications, least-cost routing (LCR) is the process of selecting the path of outbound communications traffic based on cost. Within a telecoms carrier, an LCR team might periodically (monthly, weekly or even daily) choose between routes from several or even hundreds of carriers for destinations across the world. This function might also be automated by a device or software program known as a least-cost router.
Post Dial Delay (PDD)
Post Dial Delay (PDD) is the time or delay that occurs from the time a number has been dialed until the caller or called party hears ringing.
Post dial delay is more common on wholesale termination (LCR) products, which provide aggressive rating as a product of having multiple carriers to attempt to complete the call. This occurs because each carrier can take a few seconds to acknowledge their ability to complete the call. Most carriers within the telecommunications industry consider anything under 7 seconds as an acceptable amount of PDD, and most will not troubleshoot PDD that is less than 7 seconds.
Caller identification (Caller ID) is a telephone service, available in analog and digital telephone systems, including voice over IP (VoIP), that transmits a caller‘s telephone number to the called party‘s telephone equipment when the call is being set up. The caller ID service may include the transmission of a name associated with the calling telephone number, in a service called Calling Name Presentation (CNAM). The service was first defined in 1993 in International Telecommunication Union—Telecommunication Standardization Sector (ITU-T) Recommendation Q.731.3.
The information received from the service is displayed on a telephone display screen, on a separately attached device, or on other displays, such as cable television sets when telephone and television service is provided by the same vendor.
Caller ID service is variously known by many similar terms, such as CID, calling line identification (CLI, CLID), calling number delivery (CND), calling number identification (CNID), calling line identification presentation (CLIP), and call display.
Caller ID spoofing
Main article: Caller ID spoofing
Caller ID spoofing is the practice of causing the telephone network to display a number on the recipient’s caller ID display that is different than that of the actual originating station. Many telephone services, such as ISDN PRI based PBX installations, and voice over IP services, permit the caller to configure customized caller ID information. In corporate settings this permits the announcement of switchboard number or customer service numbers. Caller ID spoofing may be illegal in some countries or in certain situations.
Caller ID spoofing is prohibited in many countries. Our company strictly adheres to the laws of the countries in which we operate and strives to suppress illegal use of Caller ID.
SkyTel passes Caller IDs for corporate clients at an additional cost, provided that the services are not used in violation of the law or used for fraudulent purposes.
Caller ID pass charges can be zero based on customer traffic.
The Caller ID pass for individuals is negotiated with your manager, for which you need to confirm the legality of using this opportunity.
Violation of the rules for passing Caller ID is punishable by a fine in the amount of 1 EUR for each minute of the call with the detected violation.
What are CC and CPS limits and what are they used for?
CC stands for “Concurrent Calls” and refers to the total number of real-time calls (including calls in call state) that you can have at the same time. A higher number of concurrent calls means a higher load on the SkyTel platform as well as on the carriers involved. Therefore, the number of simultaneous calls should always be limited to clients.
CPS stands for “Call Per Second“, which is the Maximum Rise of Calls Per Second (not to be confused with CC) and refers to the number of calls that the system or customer dialer can initiate in one second. This is especially important for customers using smart dialers. Such dialers, if not configured correctly, can place a high load on the SkyTel platform and the carriers involved with hundreds of call settings per second (for example, at the start of the day).
A standard SIP account is limited to 5 CPS and 2 CC for payphone mode, or 5 CPS and 15 CC for SIP trunk.
For more information on restrictions, please contact technical support.
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS).
The main difference between IP telephony and traditional telephony is packet switching, not circuit switching. In traditional telephony, from the moment of a call to the end of a conversation, several operators at once allocate a separate communication channel, unlike IP telephony, where packet switching is used, and the channels themselves can be used in multi-user mode.
This allows reducing the cost of a call due to more efficient use of operators’ channels, as well as flexibility in the implementation of call routing. So this allows you to improve the quality of communication to support HD, video calls, support for multi-user conferences and a large number of additional services implemented using PBX. To take full advantage of IP telephony, you need a cloud PBX service, or connect your PBX using a SIP-trunk service.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).
The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message.
SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP is designed to be independent of the underlying transport layer protocol, and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP).