Single Sign-On (SSO) enables access to multiple services with one login, simplifying user authentication and integration with various platforms.
SIP Registration is the process of establishing a connection between a device and a service provider's network, ensuring the device's legitimacy for call handling.
Real-time Transport Protocol (RTP) is used for immediate data transfer, such as in live streaming or VoIP calls. VoIP phones utilize RTP for transmitting voice data, but some firewalls might block RTP, causing audio issues in calls.
Voicemail to Email converts voicemails into digital '.WAV' files and sends them to an email address for convenient access and storage.
Skill Based Routing in call centers involves routing calls to agents based on their skills and abilities, matching customer issues with appropriate agents.
Quality of Service (QoS) in VoIP prioritizes certain data types, ensuring voice calls are transmitted efficiently. It plays a critical role in maintaining call quality by managing network bandwidth and prioritizing voice data.
SIP Trunking is a VoIP service connecting an on-site PBX to the telecom network, with the PBX managing calls and features, and the provider ensuring connection.
User Datagram Protocol (UDP) is a faster, connectionless data transfer protocol without error checking, leading to potential unreliability. Some networks drop UDP connections after a set time, causing call drops.
Voicemail Transcription uses voice-to-text technology to convert voicemails into readable text format, sent via text or email, though not always 100% accurate.
Transmission Control Protocol (TCP) is a reliable data transfer method with error checking and retransmission of lost packets, enhancing call delivery accuracy but with increased latency.
SIP authentication involves credentials provided by a service provider to validate a device's ability to place and receive calls, typically including a username and password.
Weighted Call Distribution assigns specific percentages of incoming calls to agents, suitable for low-volume call environments or varied agent skill sets.
Uniform Call Distribution routes calls to the agent who has been idle the longest, allowing efficient agents to handle more calls.
SIP Application Layer Gateway (ALG) facilitates SIP connections by modifying packet headers. While beneficial for some services, it often hinders VoIP by removing vital information, leading to issues like connection loss and call drops.
Regular Call Routing distributes calls to specified agents in a fixed order, commonly used in hierarchical situations to ensure consistent call distribution.
Session Initiation Protocol (SIP) is a standard for establishing real-time communications such as telephony and messaging. It manages initiating, maintaining, and terminating connections, ensuring smooth, direct communication between devices.
Expected Wait Time (EWT) in VoIP is a predictive metric used in call centers to estimate the waiting time for callers. While not always precise, it provides valuable insights for managing call center operations.
Do Not Disturb (DND) mode in telephony systems marks all lines as busy, directing incoming calls to voicemail, with options for different voicemail greetings and reminders to deactivate DND.
DECT, or Digital Enhanced Cordless Telecommunications, is a wireless communication standard allowing a base station connected to a service provider to communicate with cordless handsets.
Firm Order Commitment (FOC) is the confirmed date for transferring a phone number from one carrier to another, marking the end of the porting process.
Network Address Translation (NAT) in VoIP translates internal network data for internet compatibility, ensuring seamless data exchange between internal devices and external networks.
Direct Inward Dialing (DID) in VoIP allows external direct calling to specific lines. In the U.S., DIDs are essential for assigning unique e911 addresses, crucial for accurately directing emergency services.
Packet Capture (PCAP) in VoIP is a diagnostic method used by technicians to analyze data packets for troubleshooting. PCAP files contain captured data streams, aiding in network issue resolution.
Private Branch Exchange (PBX) in VoIP refers to software or hardware solutions that manage internal call routing. PBX systems assign phone numbers, manage call features, and can range from manual switchboards to modern software-based solutions.
PABX (Private Automatic Branch Exchange) is a modern PBX system that automates call switching, with most modern PBXs effectively being PABXs.
SIP Application Layer Gateway (ALG) facilitates SIP connections by modifying packet headers. While beneficial for some services, it often hinders...
Quality of Service (QoS) in VoIP prioritizes certain data types, ensuring voice calls are transmitted efficiently. It plays a critical role in...
SIP authentication involves credentials provided by a service provider to validate a device's ability to place and receive calls, typically...
Weighted Call Distribution assigns specific percentages of incoming calls to agents, suitable for low-volume call environments or varied agent skill...
Real-time Transport Protocol (RTP) is used for immediate data transfer, such as in live streaming or VoIP calls. VoIP phones utilize RTP for...