Voice over IP (VoIP)
- VoIP converts analog voice signals into digital packets that travel over IP networks instead of traditional circuit-switched telephone networks
- Voice quality depends on network conditions - latency, jitter, and packet loss directly impact call quality
- Uses Real-time Transport Protocol (RTP) for actual voice data and RTP Control Protocol (RTCP) for quality monitoring
- Requires Quality of Service (QoS) configuration to prioritize voice traffic over data traffic
VoIP Components
- IP Phones: Hardware endpoints that connect directly to network switches (Power over Ethernet capable)
- Softphones: Software applications on computers/mobile devices that function as virtual phones
- Analog Telephone Adapters (ATAs): Convert traditional analog phones to work with VoIP systems
- VoIP Gateways: Bridge VoIP networks with Public Switched Telephone Network (PSTN) for external calls
- Call Manager/PBX: Central system managing call routing, features, and user accounts
Signaling Protocols
| Protocol | Purpose | Port | Use Case |
|---|---|---|---|
| SIP (Session Initiation Protocol) | Call setup/teardown | TCP/UDP 5060 | Most common, interoperable |
| H.323 | Call setup/teardown | TCP 1720 | Legacy Cisco systems |
| MGCP | Gateway control | UDP 2427 | Service provider networks |
| SCCP/Skinny | Cisco proprietary | TCP 2000 | Cisco IP phones to Call Manager |
Network Requirements
- Bandwidth: 64-90 Kbps per call (depends on codec used)
- Latency: Must be under 150ms one-way (anything over 150ms causes noticeable delay)
- Jitter: Should be under 30ms (variation in packet arrival times)
- Packet Loss: Must be under 1% (even 3% loss severely degrades quality)
Common Codecs
| Codec | Bandwidth | Quality | Use Case |
|---|---|---|---|
| G.711 | 64 Kbps | Highest | LAN environments with plenty of bandwidth |
| G.729 | 8 Kbps | Good | WAN links with limited bandwidth |
| G.722 | 48-64 Kbps | HD Voice | Modern systems requiring high quality |
QoS Implementation
- Voice traffic requires highest priority - typically marked with DSCP EF (Expedited Forwarding)
- Call signaling traffic marked with DSCP AF31 (lower priority than voice bearer traffic)
- Use Low Latency Queuing (LLQ) to guarantee voice traffic gets processed first
- Configure traffic shaping on WAN links to prevent buffer overflow (causes jitter)
Power Considerations
- IP phones typically use Power over Ethernet (PoE) - IEEE 802.3af standard provides 15.4W
- Cisco phones may require PoE+ (802.3at) providing up to 25.5W for advanced features
- Always verify switch PoE budget - calculate total wattage needed for all connected devices
- Deploy UPS systems for switches to maintain phone service during power outages
VLAN Configuration
- Separate voice and data VLANs - prevents data traffic from impacting voice quality
- Voice VLAN typically configured as auxiliary VLAN on access ports
- Cisco Discovery Protocol (CDP) or LLDP-MED tells phones which VLAN to use
- Example: Data on VLAN 10, Voice on VLAN 20 with higher QoS priority
Notes
- VoIP is highly sensitive to network issues - problems invisible to data users become immediately apparent to voice users
- Always test voice quality under full network load conditions, not just during quiet periods
- Consider bandwidth overhead - RTP headers, IP headers, and Layer 2 headers add ~40 bytes per packet
- Jitter buffers in phones compensate for variable delay but add latency - tune appropriately for your network
- Security consideration: VoIP calls can be intercepted more easily than traditional phone calls - implement encryption (SRTP) for sensitive environments
- Emergency services (911) location - VoIP systems must be configured to provide accurate location information for emergency calls
- Backup power planning is critical - unlike traditional phones, VoIP phones don’t work when network equipment loses power
- Consider bandwidth multiplication for WAN planning - a 10Mbps link can theoretically handle 125 G.729 calls, but QoS overhead and bursting reduce this significantly