1. LOG Objectives  Describe some of the VoIP implementation challenges such as Delay/Latency, Jitter, Echo, and Packet Loss  Describe the voice encoding technique (PCM) and various compression standards  Describe the voice packetization process and the various VoIP transport protocols  Describe the elements of a VoIP Dial Plan Design  Configure Voice Interface Cards (VICs) to place VoIP Calls

2. Objective 1 Describe some of the challenges and concerns faced in VoIP implementations such as Delay/Latency, Jitter, Echo, and Packet Loss

3. Typical VoIP Deployment

4. VoIP Network

5. VoIP Deployment Challenges Three main challenges:  Delay  Jitter  Packet Loss Other Challenges: Echo, VAD, Tandem Encoding

6. Delay  Delay / Latency  4 types:  Propagation Delay  Difference in time between transmission of packet and subsequent reception  Typically measured in milliseconds or less  Handling Delay (Processing Delay)  Sampling and packetization delay at the router/switch (e.g. G.729 generates a sample every 10 ms)  Queuing Delay - Packets held in queue because of congestion on outbound interface  Serialization Delay (or Insertion Delay)  Amount of time to place the data bit / byte on an interface  Not an issue as impact is microscopic

7. Delay Budget  ITU standard states that total amount of delay in a voice call must be less than 150 ms.

8. Delay Budget

9. Jitter  Jitter  Exists only in data networks  Variation in packets inter-arrival time  Not a major issue in well designed and managed networks  Can be controlled by “jitter buffers” (or De-jitter buffers) which can statically or dynamically change size to conceal the packet arrival discrepancies

10. Packet Loss  Packet Loss  Voice is transported using UDP  Therefore, no retransmission of lost packets

11. Challenges (cont.)  VAD (Voice Activity Detection)  At least 50% of bandwidth in a typical voice call is silence  VAD basically suppresses the silence  Front-end speech clipping

12. Objective 2 Describe the voice encoding technique (PCM) and various compression standards

13. Voice Encoding  Codecs transform analog signals into a digital bit stream and digital signals back into analog signals.  Figure below shows that an analog signal is digitized with a coder for digital transport. The decoder converts the digital signal into analog form.

14. Voice Encoding (cont.)  The first basic modulation and coding technique was Pulse Code Modulation (PCM). The ITU-T standard for PCM is G.711.  With PCM, analog speech is sampled 8000 times a second. Each speech sample is mapped onto 8 bits. Thus, PCM produces (8000 samples/ second) * (8 bits/sample) = 64,000 bits/second = 64 Kbps coded bit rate.

15. Voice Coding Standards  Each codec provides a certain quality of speech.  A measure used to describe the quality of speech is the Mean Opinion Score (MOS)  With MOS, a large group of listeners judges the quality of speech from 5 (best) to 1 (bad). The scores are then averaged out to provide the MOS score for each sample.  4 is considered “Toll Quality”  3 – 4 considered Communication Quality  < 3 considered synthetic  G.711 has a MOS score of 4.1 –4.4and G.729 has a MOS score of 3.92 – 4.2  G.711 not suited for VoIP. Why?

16. Coding Standards and MOS NOTE: These Mean Opinion scores may vary

17. Module Objective 3 Describe the Voice Packetization process and the various VoIP transport protocols

18. Voice Encoding Process and Standards Digital Voice CODEC: Analog to Digital conversion in DSP Compress in DSP Analog Create Voice Datagram Add Header (RTP, UDP, IP, etc) Process Header Re-sequence and Buffer Delay Decompress CODEC: Digital to Analog Digital Analog Voice

19. Voice over IP Packet using G.729  Voice packet to be sent over IP sent in two portions:  (1) the control sent using TCP  (2) the actual packetized voice sent using RTP/UDP/IP  Packetized Voice over IP packet

20. RTP / RTCP  RTP (Real-time Transport Protocol) assists in streaming audio-video - defined in RFC 1889.  RTCP (Real-time Transport Control Protocol) controls RTP - defined in RFC 1889 Control protocol that works in conjunction with RTP. RTCP control packets are periodically transmitted by each participant in an RTP session to all other participants. Feedback of information to the application can be used to control performance and for diagnostic purposes.  Real-time traffic is carried over UDP ports ranging from 16,384 to 32,768  RTP data is transported on an even port  RTCP is carried on the next odd port.  RTCP is a session layer protocol that monitors the delivery of data and provides control and identification functions.

21. RTP / RTCP  RTP header contains a time stamp and a sequence number allowing the receiving device to buffer as much as necessary to remove jitter and latency  RTP does not address Resource Reservation  RTP does not address Quality of Service  Need for separate protocols for those tasks  QoS and RSVP

22. RTP Header