Voice Components

Voice Components

 
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Digital Circuits

Multi-Channel connectivity on a single port
Typically 4-23 channels (bearer or voice)
1 channel signalling
channel associated signalling (CAS)
common channel signalling (CCS)
T1 US, E1 Europe

Voice Codec’s

Codec
G.711 (SD)
G.722 (HD)
G.729
iLBC
iSAC
Bandwidth
64Kbps
64Kbps
8K
13.3K
10-32K
w/overhead
82K+
80K+
26K+
28K+
16-64K+
MOS
4.3/5
3.92/5
4.14/5

DSP Modules

  • Convert TDM and Analog voice calls into IP Packets
  • Sampling, encoding, and quantization
  • Transcoding
  • Conferencing

RTP

RTP uses even ports UDP range 16384-32767
RTCP offers out of band statistics and control information
SRTP is used for encrypted voice calls
Layer 2 header
IP Header
UDP Header
RTP Header
Voice Payload

VoIP Signalling Protocols

Protocol
Description
H.323
ITU standard protocol for interactive conferencing. Evolved from H.320 ISDN. Flexible Complex
MGCP
IETF standard for PSTN gateway control
SIP
IETF protocol for interactive and non-interactive conferencing. Simpler than H.323
SCCP (Skinny)
Cisco proprietary protocol used between Cisco Unified Communications Manager and Cisco VoIP phones
Latency < 150ms
Jitter < 30ms
Loss < 1%

Collaboration Call Control

Cisco Unified Communications Manager (CUCM)
  • SIP based signaling
  • VoIP and Telepresence endpoints
  • Alphanumeric / numeric dial plans
  • Cisco End Game Controller
  • ≤ 30,000 phones per cluster
  • Uses IBM Informix database
  • Customised Red Hat Kernel (not accessible)
  • GUI management and administration
 
Unified Communications Manager Express
  • Router (IOS) based call control platform
  • Fully featured IP telephony
  • Support for up to 450 phones
  • Video calling support
  • Voice conferencing
  • Advanced call routing and voice gateway
  • Licensed feature of Cisco IOS routers
 
Cisco Video Communication Server (VCS)
  • H.323 and SIP based signalling
  • Video / Telepresence endpoints
  • Firewall traversal capabilities