VOIP

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