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Telefonía IP
Arquitectura y aplicaciones - P2

Fredy Campos A.
f.campos@ieee.org

Carrera de Ingeniería Electrónica y Telecomunicaciones
Universidad Nacional Tecnológica de Lima Sur
http://www.untecs.edu.pe/portal/

ver 1.0

2015

Objetivos

● Presentar las principales arquitecturas y aplicaciones
de telefonía IP

Telefonía IP - Arquitectura y Aplicaciones - p2 2

Specifying
Requirements for
VoIP Calls

Introducing Voice over IP – Part 3

Fredy Campos
f.campos@ieee.org
NOTE:Resume extracted only for academic use

Factors Affecting Audio Clarity  Fidelity: Audio accuracy or quality  Echo: Usually due to impedance mismatch  Jitter: Variation in the arrival of voice packets  Delay: Time it takes for the signal to propagate from one end to the other end of the conversation  Packet loss: Loss of packets on the network  Side tone: Allows speakers to hear their own voice  Background noise: Low-volume noise heard at the far end of the conversation .

Jitter in IP Networks Steady Stream of Packets Time Same Packet Stream After Congestion or Improper Queuing .

Sources of Delay Packet Flow 64 kb/s Router 64 kb/s Router E1 E1 Fixed: Switch Fixed: Fixed Fixed Fixed: Dejitter Coder Delay Switch Fixed: Switch Buffer Delay Serialization Delay Delay Delay Fixed: Packetization Variable: Delay Output Queuing Delay .

Acceptable Delay: G. provided that administrators are aware 150–400 of the transmission time and its impact on the transmission quality of user applications Unacceptable for general network planning Above 400 purposes (However.) . it is recognized that in some exceptional cases. this limit will be exceeded.114 Range in Milliseconds Description 0–150 Acceptable for most user applications Acceptable.

Effect of Packet Loss Lost Audio Packet 1 Lost Packet 2 Packet 3 .

800 – Results in subjective measures – Scores from 1 (worst) to 5 (best).861 – Automated in-service measurement – Scores from 6.0 is toll quality  PSQM – Perceptual Speech Quality Measurement – Defined in ITU Standard P. 4.MOS and PSQM  MOS – Mean opinion score – Defined in ITU-T Recommendation P.5 (worst) to 0 (best) .

Voice Quality Measurement Comparison Feature MOS PSQM Test method Subjective Objective End-to-end packet loss test Inconsistent No End-to-end jitter test Inconsistent No .

12  PSTN fallback  IP RTP Priority and Frame Relay IP RTP Priority  IP to ATM class of service (CoS)  Low Latency Queuing (LLQ)  Multilink PPP (MLP)  Resource Reservation Protocol (RSVP) .QoS Mechanisms for VoIP  Header compression  Frame Relay traffic shaping (FRTS)  FRF.

Objectives of QoS  Support dedicated bandwidth  Improve loss characteristics  Avoid and manage network congestion  Shape network traffic  Set traffic priorities across the network .

Applying QoS In the Output Queue In the WAN VoIP QoS In Conjunction with IP .

37 (store-and-forward) .  Methods to transmit fax and modem over IP networks: – Terminating and transmitting the data on the gateway (fax relay) – Sending the data in-band into the RTP stream (fax pass-through) – Receiving and converting faxes to files using T. not for fax or modem data frequencies. packet loss is much more critical for fax and modem communications.Transporting Modulated Data over IP Networks  Fax and modem traffic consists of digital data modulated into high-frequency tones.  In contrast to voice.  VoIP compression algorithms are designed for voice.

711 64 kb/s G.711 64 kb/s Encoding Decoding IP Network Analog Data Tunnelled Through 64 kb/s VoIP Analog Data Analog Data 0110011 0110011 End-to-End Connection . Pass-Through Topology 0110011 0110011 G.

Fax Pass-Through Considerations  Works only when the configured codec is G.323 – SIP – MGCP .  Supported under the following call control protocols: – H.  Some gateways have limited port numbers for simultaneous use.711 or clear channel.  VAD and echo cancellation are disabled.

Modem Pass-Through Considerations

 Works only when the configured codec is G.711 or
clear-channel.
 VAD and echo cancellation need to be disabled.
 Modem pass-through over VoIP performs these functions:
– Represses processing functions
– Issues redundant packets
– Provides static jitter buffers
– Differentiates modem signals from voice and fax signals
– Reliably maintains a modem connection across the packet
network

Relay Topology
0110011 0110011
DSP DSP
Demodulates Modulates

IP Network

TCP Transmission
of Data Packets
Analog Data Analog Data

0110011 0110011

Connection 1 Connection 2 Connection 3

Fax Relay Considerations

T.38 fax relay includes these features:
 Fax relay packet loss concealment
 MGCP-based fax (T.38) and DTMF relay
 SIP T.38 fax relay
 T.38 fax relay for T.37/T.38 fax gateway
 T.38 fax relay for VoIP H.323

Modem Relay Considerations Modem relay includes these features:  Modem tone detection and signaling  Relay switchover  Payload redundancy  Packet size  Dynamic and static jitter buffers .

E-Mail Fax PSTN . Fax E-Mail PSTN  Off-ramp sends standard e-mail messages that are delivered as faxes.Store-and-Forward Fax  On-ramp receives faxes that are delivered as e-mail attachments.

Fax and Modem Pass-Through G3 Fax Original Terminating G3 Fax Initiates the Call Gateway Gateway IP Network VoIP Call T.30 CED tone Call Control Issues NSE NSE Accept Change codec Change codec VoIP Call .

Cisco Fax Relay G3 Fax Gateway Gateway G3 Fax Initiates the Call IP Network T.30 VoIP Call T.30 CED Tone DIS Message Fax Relay Switchover (PT96) Send Codec ACK (PT97) Download Codec Download Codec Codec Download Done (PT96) Codec Download ACK (PT97) Fax Relay Established .

38 UDP Packets . H.38 T.38 Channels T.30 CED Tone DIS Message Mode Request Mode Request ACK Close VoIP and Open T.30 VoIP Call T.38 G3 Fax Initiates the Call Gateway Gateway IP Network T.323 T.38 relay G3 Fax T.

30 VoIP Call T.38 UDP Packets .38 Relay G3 Fax T.38 G3 Fax Initiates the Call Gateway Gateway IP Network T.30 CED Tone DIS Message INVITE (T. SIP T.38 in SDP) 200 OK ACK T.38 T.

Gateway Signaling Protocols with Fax and Modem Pass-Through and Relay MGCP T.  Call agent-controlled mode: – Call agents instruct gateways to process fax traffic. – Allows the use of MGCP-based T.38 fax relay provides two modes of implementation:  Gateway-controlled mode: – Gateways negotiate fax relay transmission by exchanging data in SDP messages. .38 fax without the necessity of upgrading the call agent software. – Call agent can instruct gateways to revert to gateway-controlled mode if it can not handle fax control.

 DTMF relay addresses this problem.DTMF Support  DTMF tones are distorted when gateways use compression on slower WAN links. S0 256 G.729 Codec S1 kb/s Being Used 256 kb/s .

including fax and modem calls. over IP networks.Summary  Because of the nature of IP networking. delay.  Several methods may be used to determine audio quality in a VoIP network. . and jitter in a VoIP network.  QoS is used to meet the strict requirements concerning packet loss. voice packets sent via IP are subject to certain transmission problems.  There are some challenges to transporting modulated data.

and MGCP.Summary (Cont. SIP.)  These features support fax and modem traffic: – Fax and modem pass-through – Fax and modem relay – Store-and-forward fax  T. .  DTMF support is provided by Cisco IOS gateways.38 pass-through and relay use special protocol enhancements on H.323.

Understanding Codecs. and DSP Functionality Introducing Voice over IP – Part 4 Fredy Campos f.org NOTE:Resume extracted only for academic use .campos@ieee. Codec Complexity.

 Various codec standards define the compression rate of the voice payload.711 – G.  Codecs translate VoIP media streams into another format: A to D. or D to A.722 – G.726 – G.729 – G.  Supported Cisco codecs include: – G.0—1-31 .1 – GSM FR – iLBC CVOICE v6.Codecs  Codecs perform encoding and decoding on a digital data stream or signal.723.728 – G. D to D.

3 kb/s 48 16 G.723r53 5.723r63 6.711 64 kb/s 240 33 G.729 8 kb/s 20 50 G.0—1-32 .726r24 24 kb/s 60 33 G.726r16 16 kb/s 80 25 G.3 kb/s 24 33 G.726r24 24 kb/s 80 25 G.726r16 16 kb/s 40 50 G.711 64 kb/s 160 50 G.3 kb/s 20 33 CVOICE v6.726r32 32 kb/s 120 33 G.726r32 32 kb/s 80 50 G.728 16 kb/s 80 13 G.723r53 5.Impact of Voice Samples and Sample Size on Bandwidth Codec Bandwidth Sample Size Packets G.723r63 6.3 kb/s 40 17 G.729 8 kb/s 40 25 G.728 16 kb/s 40 25 G.

0—1-33 .12 Layer 2 header: 6 bytes of overhead  MP: 6 bytes of overhead CVOICE v6.Data-Link Overhead  Ethernet II: 18 bytes of overhead  PPP: 6 bytes of overhead  FRF.

Security and Tunneling Overhead  IPsec: 50 to 57 bytes  L2TP or GRE: 24 bytes  MLP: 6 bytes  MPLS: 4 bytes CVOICE v6.0—1-34 .

400 35.400 41.267 34.400 27.467 G.338 7350 13.200 G.525 G.728 16.200 87.723r63 6300 48 12.800 26.000 40 34.729 8000 40 17.000 80 25.467 G.133 47.985 G.600 G.400 19.200 39.0—1-35 .133 79.913 G.726r16 16.200 G.723r53 5300 20 17.726r24 24.000 120 44.400 11.600 G.200 17.000 240 76.600 G.600 27.200 9600 19.400 19.395 6360 12.723r53 5300 40 11.000 80 37.000 60 42.600 27.729 8000 20 26.490 7420 20.726r32 32.375 8400 21.400 67.000 160 82.200 17.711 64.670 CVOICE v6.711 64.200 G726r24 24.200 55.728 16.200 G.200 G.000 40 34.400 G.200 39.200 31.726r16 16.726r32 32.200 47.267 66.200 G.000 80 50.Calculating the Total Bandwidth for a VoIP Call Codec Frame Frame Relay Codec Sample Size Ethernet Speed Relay with cRTP Bits per Bits per Bits per Bytes Bits per Second Second Second Second G.000 80 25.723r63 6300 24 18.

490 11.267 28.395 7407 G.723r53 5300 40 11.560 G.726r16 16.000 40 34.200 16.000 80 25.400 22.728 16.800 24.Effects of VAD Frame Relay with Codec Codec Speed Sample Size Frame Relay VAD G.338 8019 G.400 22.380 G.944 G.000 60 42.200 16.760 G726r24 24.711 64.773 G.729 8000 40 17.560 G.180 G.723r63 6300 24 18.000 80 50.723r63 6300 48 12.0—1-36 .000 80 25.726r16 16.573 G.729 8000 20 26.726r32 32.000 40 34.375 11.267 49.728 16.726r32 32.400 32.400 53.000 240 76.000 160 82.360 G.726r24 24.723r53 5300 20 17.711 64.200 11.380 G.160 G.360 G.400 27.400 17.000 80 37.000 120 44.570 G.369 CVOICE v6.

 Transcoding: The conversion from one codec to another.  MTP: Two supported types on Cisco IOS routers: – Software MTPs – Hardware MTPs  Conferencing: Network-based conference bridge is required to facilitate multiparty conferences in VoIP network.  Voice termination: The digitization and packetization of an analog signal on a TDM interface.Digital Signal Processors Media resource: A software-based or hardware-based entity that performs media-processing functions on the data streams to which it is connected. CVOICE v6.0—1-37 .

0—1-38 .711 Router1 Router2 Phone1-1 Phone1-2 PSTN Phone2-1 Phone2-2 2001 2002 3001 3002 CVOICE v6. Media Resource Deployment Example San Jose Chicago Transcoding or conferencing IP WAN Conferencing G.729 IVR G.

711 (a-law and mu-law) G.729AB G. G.729B Fax Relay Fax Relay CVOICE v6.726 (all versions) G.729.0—1-39 .728 G. G.723 (all versions) G.729A. Codec Complexity Medium Complexity High Complexity (Four calls per DSP) (Two calls Per DSP) G.

medium Set codec complexity medium. secure Set codec complexity secure. lower call density. high Set codec complexity high. higher call density. Mid range complexity and call density. medium Set codec complexity medium. lower call density. Flex complexity. Cisco-router(config-voicecard)# codec complexity flex CVOICE v6. High complexity.0—1-40 . Mid range complexity and call density. <cr> Cisco-router(config-voicecard)# codec complexity high C5510 router(config)# voice-card 1 router(config-voicecard)# codec complexity ? flex Set codec complexity Flex.Configuring Codec Complexity C549 router(config)# voice-card 1 router(config-voicecard)# codec complexity ? high Set codec complexity high. High complexity.

2.2.Verifying Codec Complexity HQ-1# show voice dsp DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT ==== === == ======== ======= ===== ======= === == ========= == ===== ============ ----------------------------FLEX VOICE CARD 0 ------------------------------ *DSP VOICE CHANNELS* CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending LEGEND : (bad)bad (shut)shutdown (dpend)download pending DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ===== === == ========= ======= ===== ======= === == ========= == ==== ============ *DSP SIGNALING CHANNELS* DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ===== === == ========= ======= ===== ======= === == ========= == ==== ============ C5510 002 01 {flex} 8.0 alloc idle 0 0 0/2/0 02 0 0/0 C5510 002 02 {flex} 8.0—1-41 .0 alloc idle 0 0 0/2/1 02 0 0/0 ------------------------END OF FLEX VOICE CARD 0 ---------------------------- CVOICE v6.

and codecs: – C5510 (used on PVDM2) has higher performance than C549 (used on PVDM)  Single DSP can only be used as a single media resource: – Either voice termination.DSP Requirements for Media Resources  Number of DSPs depends on DSP type.0—1-42 . required media resources. transcoding or MTP  DSP calculator helps calculation of required DSPs: – http://www.com/cgi-bin/Support/DSP/dsp-calc. and MTP CVOICE v6. conferencing. transcoding. conferencing.cisco.pl – Support voice termination.

DSPs per PVDM2 PVDM2 Number of C5510 DSPs PVDM2-8 1/2 PVDM2-16 1 PVDM2-32 2 PVDM2-48 3 PVDM2-64 4 CVOICE v6.0—1-43 .

for example. for C5510 (PVDM2.Conferencing DSP Resources C549 (PVDM. ISR. NM-HDV) example. NM-HDV2) Maximum Participants per 6 8 Conference G.711 Conferences per 1 8 DSP Mixed-Mode Conferences 1 2 per DSP CVOICE v6.0—1-44 .

mu-law) 4 16 Sessions per DSP G.711 (a-law. for From Low Complexity To: example. GSM 4 6 EFR Sessions per DSP CVOICE v6.729A. for C5510 (PVDM2.0—1-45 .729. ISR.Transcoding DSP Resources C549 (PVDM. GSM 4 8 FR Sessions per DSP G. NM-HDV) example.729AB. G. NM-HDV2) G. G.729B.

com/cgi-bin/Support/DSP/dsp-calc.0—1-46 .pl CVOICE v6. 2 Select the Cisco IOS release. DSP Calculator URL: http://www.cisco.DSP Calculator 1 Select the router model.

) 3 Select router VICs. CVOICE v6.0—1-47 . 4 Specify the number of calls.DSP Calculator (Cont.

) 5 Specify the number of transcoding sessions.DSP Calculator (Cont.0—1-48 . CVOICE v6. 6 Specify the number of conferences.

CVOICE v6.0—1-49 .DSP Calculator Results 7 Sample Cisco IOS Calculate required configuration. DSPs or PVDMs.

Enable SCCP on the Cisco Unified Communications Manager interface or Cisco Unified Communications Manager Express 3.Configuring Conferencing and Transcoding on Voice Gateway Routers 1. Configure enhanced conferencing and transcoding CVOICE v6. Determine DSP resource requirements 2.0—1-50 .

1.201 sccp local FastEthernet 0/0 sccp local FastEthernet 0/0 sccp sccp voice-card 0 voice-card 0 dsp services dspfarm dsp services dspfarm CVOICE v6.1.1.201 Router1 Router2 Phone1-1 Phone1-2 Phone2-1 Phone2-2 1001 1002 PSTN 2001 2002 sccp ccm 10.1. DSP Farm Configuration Example San Jose Chicago Cisco Unified Communications IP WAN Manager 10.1.1.0—1-51 .201 sccp ccm 10.

0—1-52 .1. DSP Profile Configuration Example San Jose Chicago Cisco Unified Communications IP WAN Manager 10.201 Router1 Router2 Phone1-1 Phone1-2 Phone2-1 Phone2-2 1001 1002 PSTN 2001 2002 dspfarm profile 1 conference dspfarm profile 1 transcode codec g711ulaw codec g711ulaw codec g711alaw codec g711alaw codec g729ar8 codec g729ar8 codec g729abr8 codec g729abr8 codec g729r8 codec g729r8 codec g729br8 maximum sessions 6 maximum sessions 2 associate application SCCP associate application SCCP no shutdown no shutdown CVOICE v6.1.

router(config)# sccp  Enables SCCP and brings it up administratively. router(config)# sccp local <interface>  Selects the local interface that SCCP applications use to register with Cisco Unified Communications Manager.0—1-53 . CVOICE v6. DSP Farm Configuration Commands for Enhanced Media Resource router(config)# sccp ccm {ip-address | dns} identifier identifier-number [port port-number] [version version-number]  Adds a Cisco Unified Communications Manager server to the list of available servers to which the Cisco voice gateway can register.

DSP Farm Configuration Commands for Enhanced Media Resource (Cont. MTP.0—1-54 . CVOICE v6.) router(config)# voice-card slot  Enters the voice card configuration mode. or transcoding. router(config)# dspfarm profile profile-identifier {conference | mtp | transcode}  Creates a DSP farm profile for conferencing. router(config-voicecard)# dsp services dspfarm  Enables DSP farm services.

router(config-dspfarm-profile)# associate application SCCP  Enables SCCP for the profile. DSP Farm Configuration Commands for Enhanced Media Resource (Cont.0—1-55 . router(config-dspfarm-profile)# maximum sessions number  Defines the maximum number of sessions. CVOICE v6.) router(config-dspfarm-profile)# codec codec-type  Specifies the allowed codecs.

Maximum Packetization Period : 30 .0—1-56 . Transcoder: Not Required Codec : g729abr8. Resource ID = 1 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED DSP farm profile active Resource Provider : FLEX_DSPRM Status : UP and associated with SCCP Number of Resource Configured : 2 Number of Resource Available : 2 Codec Configuration Codec : g711ulaw. Maximum Packetization Period : 60 . Transcoder: Not Required Codec : g711alaw. Maximum Packetization Period : 60 . Service = CONFERENCING. Maximum Packetization Period : 60 . Transcoder: Not Required CVOICE v6. Maximum Packetization Period : 30 . Transcoder: Not Required Codec : g729ar8. Transcoder: Not Required Codec : g729r8. Maximum Packetization Period : 60 .Verifying Media Resources Router# show dspfarm profile 1 Dspfarm Profile Configuration Profile ID = 1. Transcoder: Not Required Codec : g729br8.

- 0 5 1. Verifying Media Resources (Cont.0—1-57 . .0.0.6 UP N/A FREE conf 1 . - Total number of DSPFARM DSP channel(s) 2 Two conference bridges configured CVOICE v6. .) Router# show dspfarm dsp all SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED 0 5 1.6 UP N/A FREE conf 1 .

data-link overhead. CVOICE v6. and RTP have positive and negative impacts on total bandwidth. sample size.  Codec choice.  Voice sample size is a variable that can affect the total bandwidth used.0—1-58 .  Codec complexity affects the call density.Summary  Codecs are used to compress and decompress various types of data that would otherwise use up large amounts of bandwidth.  Several factors must be included in calculating the overhead of a VoIP call.

)  DSPs enable Cisco platforms to efficiently process digital voice traffic.  DSP farm services are enabled on the voice card. CVOICE v6.  The configuration of transcoding and conferencing on a voice gateway involves several components.  You may verify DSP media resources using show dspfarm commands.  The number of DSPs required is a key factor when deploying media resources using DSPs.0—1-59 .Summary (Cont. and DSP profiles create the actual media resource.

Fredy Campos A.org Carrera Profesional de Ingeniería Electrónica y Telecomunicaciones Universidad Nacional Tecnológica del Cono Sur de Lima http://www.p2 60 .untecs. f.Arquitectura y Aplicaciones .edu.pe/portal/ Telefonía IP .campos@ieee.