How are phone numbers used for voice calls?

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Phone numbers are the primary means by which users initiate and receive voice calls across global telecommunication networks, whether they are traditional landlines, mobile phones, or Voice over IP (VoIP) services. The process of using a phone number to establish a voice call involves a complex interplay of signaling protocols, network databases, and switching equipment.

Here's a breakdown of how phone numbers are used for voice calls, covering different scenarios:

1. Traditional Landline (PSTN) Voice Calls:
In the Public Switched Telephone Network (PSTN), phone numbers are inherently tied to physical copper lines and switching equipment.

Dialing: When you dial a phone number from a landline phone, the digits are converted into electrical signals (tones or pulses) and sent to your local telephone exchange (also called a Central Office).
Number Analysis and Routing: The local exchange analyzes the dialed phone number (which follows the E.164 standard).
Local Call: If the number belongs to a subscriber connected to the same exchange, the exchange directly routes the call to that subscriber's line, causing their phone to ring.
Long-Distance/Inter-Exchange Call: If the number belongs to a different exchange (even within the same city), the call is routed via higher-level Tandem Offices or Toll Offices through romania phone number list trunk lines (high-capacity communication paths) to the appropriate destination exchange. Each switch in the path uses routing tables based on the phone number's structure (country code, area code, local prefix) to determine the next hop.
Circuit Establishment: As the call is routed, a dedicated circuit (a continuous physical or virtual connection) is established between the calling and called parties for the duration of the conversation. This is known as circuit-switched communication.
Connection: Once the circuit is complete to the destination exchange, the called phone rings. When answered, the circuit is fully connected, and voice communication begins.
2. Mobile Voice Calls (GSM/3G/4G VoLTE/5G VoNR):
Mobile networks add layers of complexity due to subscriber mobility, but phone numbers remain the user-facing identifier.

Mobile Originated Call:
Dialing: You dial a phone number on your mobile device.
Signaling to MSC: Your phone sends a call setup request (containing the dialed MSISDN – your contact's phone number) to the nearest cell tower, which forwards it to the serving Mobile Switching Center (MSC).
Number Translation (if needed): For calls to other mobile networks or landlines, the MSC may perform a Number Portability (NP) lookup (an LRN dip) to a central database. This determines the actual network where the dialed number currently resides, as numbers can be ported between carriers.
Routing to Destination Network: Using SS7 signaling (for 2G/3G) or SIP (for 4G VoLTE/5G VoNR within an IP Multimedia Subsystem - IMS), the call is routed through the originating carrier's Gateway MSC (GMSC) to the GMSC of the destination network.
Mobile Terminated Call:
HLR Query: The destination GMSC queries the Home Location Register (HLR) of the dialed phone number's subscriber. The HLR stores the subscriber's permanent data and their current location (i.e., which Visitor Location Register - VLR - they are currently registered with).
VLR Query and Paging: The HLR then queries the indicated VLR to get the precise location of the mobile device (e.g., the specific MSC/Location Area it's in). The serving MSC then initiates a paging process, sending messages to cell towers in that Location Area to locate the device.
Radio Channel Setup: Once the device responds, a dedicated radio channel (air interface) is established between the cell tower and the mobile phone.
Call Connection: The voice path is then connected, traversing the mobile network's core (via circuit-switched paths for 2G/3G or IP packets for VoLTE/VoNR) and potentially interconnecting with the PSTN if the caller/receiver is on a landline.
3. VoIP Voice Calls:
VoIP services use phone numbers to bridge between the IP world and traditional telephony.

Mapping to SIP URI: When you dial a phone number from a VoIP client (softphone, IP phone), the number is typically translated into a SIP URI (e.g., sip:+[email protected]).
SIP Signaling: The VoIP client sends a SIP INVITE message to its VoIP service provider's SIP Proxy Server.
Routing Decisions: The SIP Proxy Server analyzes the dialed number.
IP-to-IP Call: If the destination is another VoIP user on the same or an interconnected IP network (and can be resolved via ENUM or internal directories), the call stays entirely on the IP network, with SIP managing the session setup.
IP-to-PSTN Call: If the destination is a traditional landline or mobile phone, the SIP Proxy routes the call to a PSTN Gateway.
Gateway Translation: The PSTN Gateway converts the SIP signaling and voice packets into the appropriate SS7/PSTN signaling and circuit-switched voice format, effectively translating the VoIP call to a traditional phone call for routing through the PSTN.
Voice Transmission (RTP): For the actual voice data, VoIP uses the Real-time Transport Protocol (RTP) to send voice packets over the internet.
In all these scenarios, the phone number (specifically the E.164 compliant MSISDN) serves as the human-readable address that initiates a complex process of signaling, database lookups, and network routing to establish a real-time voice connection between two parties, regardless of the underlying network technology.