The sources define the VoIP architecture as a converged communications system built upon three core interconnected components: the IP PBX, SIP Trunking, and the critical underlying network infrastructure, all managed by explicit Quality of Service (QoS) mechanisms,.

Core VoIP Architecture Components

1. IP PBX (Internet Protocol Private Branch Exchange) The IP PBX is the "brain" or central nervous system of the modern business phone setup,. It uses Internet Protocol (IP) and packet switching technology to handle internal and external calls over the organization's existing internet connection, moving away from the circuit-switching used by traditional PBX systems,,,,.
   
   

   • Function: It manages call processing, routing rules, and advanced features such as conferencing, voicemail, and remote user functionality,,. The IP PBX hosts the softswitch capability and manages the IP-enabled endpoints, like VoIP phones and softphone clients.

   • Components: Key parts of the system include the Call Control Server (the brain), Media Server (handling audio processing and voicemail), Endpoints (IP phones/softphones), Gateways (for connection to traditional networks), and the LAN/WAN Integration.

   • Deployment: The architecture can be deployed on-premise (owning the hardware/software) or hosted/cloud-based (the provider maintains the PBX in a data center),.

2. SIP Trunking and SIP Protocol SIP Trunking provides the virtual "lines" or connectivity necessary for the IP PBX to communicate with the outside world and the Public Switched Telephone Network (PSTN) over the internet,,,,. A "trunk" consolidates multiple virtual communication channels, often exceeding 20, into a single logical connection,.
   
   

   • SIP (Session Initiation Protocol) is an application-layer signaling protocol that defines the rules for initiating, modifying, and terminating real-time communication sessions, including voice, video, and messaging,,,.

   • Signaling vs. Media: A critical architectural distinction is the separation between the control plane and the media plane.

     • SIP manages the signaling (call setup, using commands like INVITE, ACK, and BYE),,. SIP signaling is bursty and requires guaranteed delivery for reliable call setup and tear-down.

     • Real-time Transport Protocol (RTP) is the separate protocol used to transmit the actual audio or video media stream once the session is live,,. RTP traffic is continuous and highly sensitive to delay.

3. Endpoints and Network Devices The system relies on Endpoints (such as IP phones, softphones, and mobile apps) for user access,. Crucially, the VoIP architecture is dependent on Network Infrastructure Devices like routers and switches, which are responsible for enforcing QoS policies,. Session Border Controllers (SBCs) are specialized components that sit between the network and the outside world, managing Quality of Service (QoS) and providing security and proper bandwidth allocation for communications,.
   
   

QoS in the Converged Network Context

The shift to IP telephony requires mandatory expertise in QoS because voice performance relies entirely on the stability and configuration of the data network (LAN/WAN). High latency, jitter, and packet loss are secondary symptoms caused by network congestion and resource contention.

1. Prioritization through Marking

QoS in a converged network starts with classifying and marking VoIP traffic so network devices can identify and prioritize it,.

Traffic Type

Protocol

QoS Classification (Layer 3)

QoS Classification (Layer 2)

Priority Goal

Real-Time Media

RTP

DSCP 46 (Expedited Forwarding - EF),,,

802.1p CoS 6 (Voice/Critical),,,,

Highest priority for low latency/loss

Signaling

SIP

DSCP 24 or 34 (Assured Forwarding - AF),

802.1p CoS 3 or 4

Guaranteed delivery for call setup

The IP phone or softphone must be correctly configured to tag its outbound traffic with DSCP 46. DSCP marking uses 6 bits of the IP Header's ToS field to provide up to 64 classes for traffic,.

2. Layer 2 QoS Implementation (Switches)

Layer 2 switches are critical for granular QoS enforcement within the Local Area Network (LAN). Since Layer 2 switches cannot natively read the Layer 3 DSCP field, they must be configured for DSCP Trust Mode,.

• When trust mode is enabled, the switch inspects the DSCP value (e.g., 46) and maps it to the corresponding 802.1p Class of Service (CoS) priority (e.g., 6),.

• The switch then uses this CoS value to place the voice traffic into a dedicated, high-priority output queue, typically using Strict Priority Queuing (SP) to ensure voice packets are transmitted immediately ahead of standard data traffic,.

• Using a dedicated Voice VLAN for all voice traffic is also a best practice, enabling granular control and traffic shaping rules specific to real-time communication,.

3. Critical QoS Metrics for VoIP

Successful operation of the VoIP architecture is measured against strict performance thresholds to ensure acceptable user experience (Quality of Experience, QoE),,:

• Latency (Delay): The one-way delay must be less than 150 ms for real-time voice traffic to ensure quality comparable to the PSTN,,,.

• Jitter (Delay Variation): The variation in packet arrival times should ideally be less than 30 ms for excellent voice quality,,,,.

• Packet Loss: The percentage of voice packets lost should be less than 1% to avoid noticeable audible errors,,,.

• Mean Opinion Score (MOS): This subjective scale, rated from 1 (poor) to 5 (excellent), is used to assess overall audio quality,,. A MOS score of 4.0 or above is generally considered high quality and is the target for professional systems,,,. High network congestion directly causes the MOS score to degrade,.

For example, high-volume data transfers (like a 23 terabyte PowerPoint presentation) that saturate a link can cause delay and dropped packets for VoIP traffic if prioritization is not properly configured, resulting in call quality degradation. To guarantee QoS across links that cannot natively honor Layer 2 markings (like a VPN/WAN link), QoS Mapping is used to convert 802.1p tags to DSCP tags for transit, and then convert them back upon arrival at the next 802.1p-capable segment,.