Voice Service using Internet Protocol over best effort broadband internet, Public IP carrier or Private IP Network
Lower Cost for
Comparision of voice over PSTN and voice over IP Low and variable High Quality of service 200-700ms <100ms Latency Dynamically allocated Dedicated Bandwidth Packet switched Circuit switched Switching Voice over IP Voice over PSTN Concept
Quality of Service (QoS) is about bandwidth (and latency) management.
IAB concerns affected by persistent, high packet drop rates that would arise from rapid growth in best effort telephony on best effort networks
Example of potential for trouble Georgia,Atlanta Nairobi, Kenya Access Links 128 Kbps last hop bottleneck
TCP data traffic(1500B packet size) + VoIP traffic produces additional 90 ms delay for some packets which is above ITU-T recommended time for speech traffic.
Occur in networks with flows that traverse multiple congested links having persistent, high packet drop rates.
All traffic slows to a crawl and nobody gets acceptable packet delivery or acceptable performance.
Use end to end congestion control
Call rejection (not available for best-effort traffic)
User Quality 2 1 N 1 2 N 64 Kbps 64 Kbps 128 Kbps Arrival Rate = N * 64 Kbps Successful transmission = 2 * 64 Kbps Drop rate = (N-2) /N
Drops occur randomly, and none of the flows can be expected to present better quality service to users.
End-to-end congestion control be used by each VoIP, and use a codec that can adapt the bit rate to the bandwidth actually received by that flow.
Rule of thumb
When packet loss rate > 20 %
Audio quality of VoIP is degraded beyond usefulness due to bursty nature of traffic.
Considering TCP flows sharing connection with VoIP flows, VoIP crowds out TCP.
Allocate bandwidth on congested links to classes of traffic.
VoIP traffic should not be exempt from end to end congestion control.
Congestion control for real time traffic
Current effort in IETF
Adaptive rate Audio Codecs
RTP (RFC 3551)
Supports the transport of real-time media, including voice traffic, over packet networks which suppresses silence conditions .
Contain media information and a header, providing information to the receiver that allows the reordering of any out-of-order packets.
Uses payload identification to describe the encoding of the media so that it can be changed in light of varying network conditions.
Encoding Scheme for audio
Sample based encoding (DVI4, G722)
Frame based encoding (G723, GSM)
In order to carry RTP in protocols offering a byte stream abstraction, such as TCP the application MUST define its own method of delineating RTP and RTCP packets.
RTP data SHOULD be carried on an even UDP port number and the corresponding RTCP packets SHOULD be carried on the next higher (odd) port number.
No specification of security services. Possibility of DOS attack for data encodings using compression techniques that have non-uniform receiver-end computational load.
Lower variation of throughput over time compared to TCP, thus suitable for applications such as telephony or streaming media.
Designed for applications that use a fixed packet size, and vary their sending rate in packets per second in response to congestion.
TFRC-PS is a variant of TFRC for applications having a fixed sending rate but vary their packet size in response to congestion.
Specified for unreliable flows, with the application being able to specify either TCP-like or TFRC congestion control.
Congestion control identifiers
CCID 2 for TCP-like congestion control
CCID 3 for TFRC congestion control.
CCID 4 for TFRC-PS congestion control (future use).
Adaptive Rate Audio Codecs
Operates at a low sending rate, or reduces the sending rate as throughput decreases and/or packet loss increases.
Improves the scalability of VoIP or TCP sharing a congested link
Effective use of available bandwidth.
supports eight speech encoding modes having bit rates between 4.75 and 12.2 kbps.
Adaptive Rate Audio Codecs
Reduces transmission rate during silence periods.
Supports audio from different channels to be separately encoded and decoded each of the individual channels.
Unequal Bit-error Detection and Protection.
Employs forward error correction (FEC) and frame interleaving, to increase robustness against packet loss.
MGCP (Media Gateway control Protocol)
System Control Unit
Provides call control and framing capabilities.
Includes following standards
Audio and Video Codecs
Define the format of audio and video information and represent the way audio and video are compressed and transmitted over the network .
Required audio and video codecs
G.723 and H.263 are default codecs preferred for NetMeeting connections, which offer the low-bit rate connections necessary for audio and video transmission over the Internet.