~/netref / VoIP & Collaboration
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VoIP & Collaboration

Carrying real-time voice over IP — signaling, media, codecs and the QoS that keeps calls clear.

// components
IP PhoneEndpoint — registers to the call server
IP PBX / Call AgentCUCM, Asterisk — routes calls & adds features
Voice GatewayBridges VoIP to the PSTN / analog lines
GatekeeperH.323 admission control & address translation
SBCSession Border Controller — security at the SIP edge

// voice QoS targets

  • One-way latency < 150 ms (ITU G.114)
  • Jitter < 30 ms — de-jitter buffer smooths arrival
  • Packet loss < 1% (PLC conceals small gaps)
  • Mark RTP DSCP EF (46), signaling CS3 (24)
  • ~87 kbps per G.711 call including L2/L3 headers
// signaling & media protocols
SIPTCP/UDP 5060 · 5061 TLSSession signaling — sets up & tears down calls
H.323TCP 1720Legacy ITU signaling suite
RTPUDP 16384+Carries the actual voice / video media
RTCPUDP (RTP+1)Reports media quality — jitter, loss
SDPinside SIPNegotiates codecs, ports & media
MGCPUDP 2427Centralized gateway control
SCCPTCP 2000Cisco “Skinny” phone protocol
// codecs
CodecBitrateMOS
G.71164 kbps4.1
G.7298 kbps3.9
G.72264 kbps4.5
Opus6–510 kbps4.5
iLBC15 kbps4.1

// SIP call flow

CALLERCALLEE
INVITE
▶
100 Trying
◀
180 Ringing
◀
200 OK
◀
ACK
▶
◀ RTP media stream (UDP) ▶
BYE
▶
200 OK
◀

// essentials

  • SIP signals calls (5060 / 5061 TLS); RTP carries media over UDP
  • RTCP reports media quality alongside RTP
  • Codecs: G.711 (64 kbps, clear) vs G.729 (8 kbps, compressed)
  • Voice VLAN + DSCP EF (46) protect call quality
  • MOS score (1–5) rates perceived call quality