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Voice over IP Fundamentals

CHAPTER 6 + 7 Routing Protocols VoIP: An In-Depth Analysis. Voice over IP Fundamentals. OSI Reference Model Open System Interconnection Model Seven Layered Model Developed by the International Standards Organization Predated by the TCP/IP Model. OSI Layers: Application

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Voice over IP Fundamentals

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  1. CHAPTER 6 + 7 • Routing Protocols • VoIP: An In-Depth Analysis Voice over IP Fundamentals

  2. OSI Reference Model • Open System Interconnection Model • Seven Layered Model • Developed by the International Standards Organization • Predated by the TCP/IP Model

  3. OSI Layers: • Application • Provides Services to applications • E-Mail • Web Browsing • Word Processing • Presentation • Formats Data • Encryption • Compression • ASCII, EBCDIC

  4. OSI Layers: • Session • Establishes, Manages and Terminates Sessions between applications • Dialog Control • Transport • Ensures Reliable Transport of Data • Transmission Control Protocol (TCP) • User Datagram Protocol (UDP) • Reliable Transport Protocol (RTP) • Port Numbers

  5. OSI Layers: • Network • Packet Formatting • Logical Addressing • Routing • Data Link • Provides reliable transport across a physical link • Physical Addressing • Media Access Control (MAC)

  6. OSI Layers: • Physical • Converting data to physical impulses • EIA/TIA-232 • V.35 • RS-449 • 802.3 • Others

  7. Addressing: • Physical (MAC) • 48 bit Hexadecimal Address burned into device memory • 24 bits Organizational Unique Identifier (OUI) • 24 bits Serial Number • Layer 2 of the OSI Model

  8. Addressing: • Logical (IP,IPX,AppleTalk) • IP Most common • IPv4 (32 bits) • IPv6 (128 bits) • Dotted Decimal Format (IPv4) • Classes A, B, C, Multicast, and Expirmental • Class A: 0.0.0.0 through 127.255.255.255 • Class B: 128.0.0.0 through 191.255.255.255 • Class C: 192.0.0.0 through 223.255.255.255 • Multicast: 224.0.0.0 through 239.255.255.255 • Experimental: 240.0.0.0 through 255.255.255.254

  9. Connecting to the Network:

  10. Routing Protocols: • Distance-Vector Routing • View from directly connected neighbors • BGP • EIGRP • RIP • Link-state Routing • View of entire network • IS-IS • OSPF

  11. Using Virtual LAN’s (VLAN’s) to Subdivide Switch: • A VLAN = a Broadcast Domain = An IP Subnet • A virtual division of the switch. • Routers are used to interconnect VLAN’s • Benefits: • Increased performance • Improved manageability • Physical topology independence • Increased security

  12. Switch Trunking: • Switches can be interconnected via a single connection • Uses either IEEE 802.1Q (Standard) or Inter-Switch Link protocol (ISL) a Cisco proprietary. • Native VLAN carries all management information • All frames are “Tagged” to cross the trunk link except for the native VLAN frames. • Tagging adds bits onto frame which are removed prior to exiting the switch on any line not a trunk • Tagging adds delay • Tagging saves physical ports • VLAN’s are distributed to all switches via Virtual Trunking Protocol (VTP)

  13. Virtual Trunking Protocol: • Switches exchange VLAN information automatically • VTP Domain Names and passwords are case sensitive • VTP Modes are server, client or transparent • VTP Server allows the creation or deletion of VLAN’s throughout system. VLAN information is saved in switch memory • VTP Client allows only the acceptance of VLAN’s from the server. Information is not stored in memory. • VTP Transparent mode allows the creation or deletion of VLAN’s of local significance only. VLAN information is stored in switch memory. Will pass VTP information to other switches within the same domain.

  14. Virtual Trunking Protocol continued: • Each VTP change increased VTP revision number. Highest revision number is distributed through out system • Configuration: • Switch(config)#vtp mode server • Switch(config)#vtp domain PHONE_NETWORK • Switch(config)#vtp password VOICEPA55 • Switch(config)#end • Switch#

  15. Virtual Trunking Protocol continued: • Interface trunking modes: • Dynamic desirable: Cisco default. Will become trunk depending on mode and device attached. • Dynamic auto: Will become a trunk depending on mode and device attached but will not actively try to negotiate a trunk link. • Trunk: Will be in trunk mode but will negotiate with either dynamic auto, dynamic desirable using Dynamic Trunking Protocol (DTP). • Access: Not in a trunk mode. Gives access to one Data VLAN and one Voice VLAN only. • Nonegotiate: Disables DTP messages on interface

  16. Creating VLAN’s on a Switch: Switch(config)#vlan 10 Switch(config-vlan)#name DATA Switch(config-vlan)#vlan 50 Switch(config-vlan)#name VOICE Switch(config-vlan)#exit Switch(config)#int fa0/1 Switch(config-if)#switchport trunk encap dot1q Switch(config-if)#switchport mode trunk Switch(config-if)#switchport trunk native vlan 1 Switch(config-if)#int fa0/2 Switch(config-if)#switchport mode access Switch(config-if)#switchport access vlan 10 Switch(config-if)#switchport voice vlan 50 Switch(config-if)#end Switch#

  17. Creating Trunk Ports on a Router: Router(config)#int fa0/0 Router(config-if)#no shut Router(config-if)#int fa0/0.10 Router(config-subif)#encapsulation dot1q 10 Router(config-subif)#ip address 1.10.0.1 255.255.255.0 Router(config-subif)#ip helper-address 172.16.2.5 Router(config-subif)#int fa0/0.50 Router(config-subif)#encapsulation dot1q 50 Router(config-subif)#ip address 1.50.0.1 255.255.255.0 Router(config-subif)#ip helper-address 172.16.2.5 Router(config-subif)#int fa0/0.1 Router(config-subif)#encapsulation dot1q 1 native Router(config-subif)#ip address 1.1.0.1 255.255.255.0 Router(config-subif)#end Router#

  18. Creating Dynamic Host Control Protocol (DHCP) on a Router: Router(config)#ip dhcp pool DATA Router(dhcp-config)#network 1.10.0.0 255.255.255.0 Router(dhcp-config)#default-router 1.10.0.1 Router(dhcp-config)#dns-server 4.2.2.2 Router(dhcp-config)#ip dhcp pool VOICE Router(dhcp-config)#network 1.50.0.0 255.255.255.0 Router(dhcp-config)#default-router 1.50.0.1 Router(dhcp-config)#dns-server 4.2.2.2 Router(dhcp-config)#option 150 ip 1.50.0.1 Router(dhcp-config)#exit Router(config)#ip dhcp excluded-address 1.10.0.1 Router(config)#ip dhcp excluded-address 1.50.0.1 Router(config)#end Router#

  19. IP Phone Boot: IP Phone connects to switchport Switchport senses and supplies PoE Via CDP phone receives voice VLAN information Phone sends DHCP request on voice VLAN and receives IP address, Mask and default-Gateway Once addressed the phone contacts TFTP server (Option 150) and downloads configuration files Phone contacts first call processing center (CME Router) and registers. If unable to contact will contact additional centers as listed in configuration

  20. Network Time Protocol (NTP): • Assigns correct date and time to voice mail • Displays correct date and time on phone • Synchronizes system • Router(config)#ntp server 64.209.210.20 • Router(config)#clock timezone WARWICK -5 • Router(config)#clock summer-time EST recurring 2 Sunday March 02:00 1 Sunday November 02:00 • Router(config)#end • Router#

  21. Network Time Protocol (NTP) continued: Router#show ntp associations Router# show clock

  22. Network Delay: Propagation Delay Handling Delay Queuing Delay * Total acceptable delay is 150 mSec

  23. Jitter: • Variation in delay affecting packet arrival time

  24. Converting Analog to Digital: • Sample the signal • Quantize the signal • Encode the quantized value into binary format: • Optionally compress the sample to save bandwidth.

  25. Sample the Signal: • How often to Sample? • Nyquist – 18,000 Samples per second! • Realistically to recognize voice and mood 8,000 Samples per second. • Result less quality less bandwidth • Process referred to as Pulse Amplitude-Modulation (PAM)

  26. Quantize the Signal: • How many Digits? • Known as Quantization • Divided into sixteen (16) segments. 0 through 7 positive and 0 through 7 negative • Values are not evenly spaced to allow for more accurate recreation of voice patterns

  27. Encode the Quantized Signal: • How many Digits? • Each Quantized value is encoded into an eight bit (8) binary number. • Total bandwidth is equal to eight bits for each sample times eight thousand samples per second. • 8 X 8000 = 64Kbps

  28. Compress the Sample: • Why? • Save bandwidth. • Reduces quality of voice • As low as 8Kbps

  29. Converting Analog Voice to Digital: • The average human can hear frequencies of 20-20,000 Hz • Human speech uses frequencies from 200-9000 Hz • Telephone channels typically transmit frequencies of 300-3400 Hz • The Nyquist theorem is able to reproduce frequencies of 300-4000 Hz

  30. Converting Analog Voice to Digital continued: • Sample at twice the highest frequency to reproduce accurately (Nyquist) • Quantization is the term used to describe the process of converting an analog signal into a numeric quantity • Since an eight (8) bit binary number can represent a value from zero (0) through two-hundred fifty-five (255) we use the Most Significant Digit (MSD) to represent positive/negative value • A zero (0) in the MSD represents a positive (+) value • A one (1) in the MSD represents a negative (-) value • The result is a range of zero through positive one-hundred twenty-seven (0 through +127) and negative one through negative one-hundred twenty-seven (-1 through -127) • Answer: -76

  31. Converting Analog Voice to Digital continued: • Codec’s convert Analog voice into Digital transmissions. • Different Codec’s convert in different methods with more or less complexity • Available Codec’s: • G.711 • Internet low Bitrate Codec (iLBC) • G.729 • G.726 • G.729a • G.728 • Is the Codec supported in the system • How many Digital Signal Processors (DSP’s) are used

  32. Converting Analog Voice to Digital continued: • Does the Codec meet satisfactory quality levels • How much bandwidth does the Codec consume • How does the Codec handle packet loss • Does the Codec support multiple sample size

  33. Codec’s: • Codec Bandwidth MOS • Consumed • G.711 64 Kbps 4.1 • Internet Low 15.2 Kbps 4.1 • Bitrate Codec (ilBC) • G.729 8 Kbps 3.92 • G.726 32 Kbps 3.85 • G.729a 8 Kbps 3.7 • G.728 16 Kbps 3.61 • MOS (Mean Opinion Score) is determined by listeners listening to the phrase “Nowadays, a chicken leg is a rare dish.” and scoring the quality of the connection on a one to five scale.

  34. Calculating Total Bandwidth Needed per Call: • Determine sample size: A larger sample is more efficient (Example: 30 bytes of voice to 50 bytes of overhead 30/80x100%=37.5% is Voice)(Example: 20 bytes of voice to 50 bytes of overhead 20/70x100%=28.5% is voice) • A larger sample takes longer to prepare, so in circuits with delay the voice call will not be as good. • Bandwidth can be saved using Voice Activity Detection (VAD) where no packets are sent during a time when there is no voice • VAD can account for 35-40% of total call time • RTP header compression does not repeat the header after the first packet since the information will stay the same for the length of the call saving 40%

  35. Calculating Total Bandwidth Needed per Call continued: • Determine CODEC used • Determine sample size • Determine layer overhead • Layer 2 datalink • Ethernet: 20 bytes • Frame-Relay: 4-6 bytes • Point-to-point Protocol (PPP): 6 bytes • Layer 3 and 4, network and transport • IP: 20 bytes • UDP: 8 bytes • Real-time Transport Protocol (RTP): 12 bytes • Typically layers 3 and 4 are always 40 bytes

  36. Calculating Total Bandwidth Needed per Call continued: • Bytes-per-packet = (Sample_size * Codec_bandwidth) / 8 • Total_bandwidth = Packet_size * Packets_per_second • Add any additional overhead: • GRE/L2TP: 24 bytes • MPLS: 4 bytes • Ipsec: 50-57 bytes • Call A: Call B: • 30 mSec Sample size 20 mSec Sample size • G.711 Codec G.729 Codec • Ethernet network Frame-relay network (4 byte)

  37. Calculating Total Bandwidth Needed per Call continued: • Call A: • (.03 * 64Kbps) = 1.92Kbps / 8 = 240 bytes • 240 + 20 (ethernet) + 40 (layer 3 and 4) = 300 bytes • 300 * (1 / .03) = 10K bytes per second • 10K * 8 = 80Kbps • Call B: • (.02 * 8Kbps) = 160bps / 8 = 20 bytes • 20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes • 64 * (1 / .02) = 3.2K bytes per second • 3.2K * 8 = 25.6Kbps

  38. Calculating Total Bandwidth Needed per Call Compared continued: • Call B: G.729 • (.02 * 8Kbps) = 160bps / 8 = 20 bytes • 20 + 4 (frame-relay) + 40 (layer 3 and 4) = 64 bytes • 64 * (1 / .02) = 3.2K bytes per second • 3.2K * 8 = 25.6Kbps • Call B: G.711 • (.02 * 64Kbps) = 128Kbps / 8 = 160 bytes • 160 + 4 (frame-relay) + 40 (layer 3 and 4) = 204 bytes • 204 * (1 / .02) = 10.2K bytes per second • 10.2K * 8 = 81.6Kbps • Savings of 68.6% using the G.729 Codec!

  39. Digital Signal processors: • DSP’s perform the function of sampling, encoding, and compression of all audio signals coming into the router. • DSP’s might be located on the routers motherboard • DSP’s might also be add on modules similar to SIMM memory modules on the motherboard called Packet Voice DSP Modules (PVDM) • DSP modules can contain multiple DSP circuits • PVDM2-8: Provides .5 DSP chip • PVDM2-16: Provides 1 DSP chip • PVDM2-32: Provides 2 DSP chips • PVDM2-48: Provides 3 DSP chips • PVDM2-64: Provides 4 DSP chips • Codec’s G.711 (a-law and u-law) (u-law is United States, Japan) (a-law All others), G.726, G.729a, and G.729ab are all of medium complexity • Codec’s G.728, G.723, G.729, G.729b and iLBC are all high complexity

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