Understanding Voice over IP Protocols Cisco Systems Service Provider Solutions Engineering February, 2002 1
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 2
Why Move to VoIP? Cost savings toll bypass Open standards H.323, SIP, MGCP Multi-vendor interoperability Integrated IP voice and data networks 3
Cisco Packet Voice Architecture TDM/ Circuit Switch Line Concentration Digital Trunk Subsystem Switching Network Call Control Connection Control Features Common Channel Signaling Complex Administration Maintenance Billing Open Service Application Layer (JAIN, AIN, TAPI, JTAPI, XML etc.) Open/Standard Interface Open Call Control Layer (SIP, H.323, MGCP, etc.) Open/Standard Interface Standards-Based Packet Infrastructure Layer (IP, ATM) 4
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 5
Early Adopters Advanced Services and Toll-Bypass Regulatory opportunities allowed for toll-bypass PC-to-phone, calling-card and international fax services Cisco-based carriers used standard protocols, but not all carriers implemented standards Inter-carrier connections had protocol interoperability challenges 6
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 7
Making the Rules for VoIP IETF (Internet Engineering Task Force) The community of engineers that standardizes the protocols that define how the Internet and Internet Protocols work. http://www.ietf.org/ ITU (International Telecommunications Union) An international organization within the United Nations System where governments and the private sector coordinate global telecom networks and services. http://www.itu.int/home/index.html 8
Defining the VoIP Protocols H.323 SIP MGCP H.248 Megaco An ITU Recommendation that defines Packet-based multimedia communications systems. H.323 defines a distributed architecture for creating multimedia applications, including VoIP Defined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia applications, including VoIP Defined as IETF RFC 2705. MGCP defines a centralized architecture for creating multimedia applications, including VoIP An ITU Recommendation that defines Gateway Control Protocol. H.248 is the result of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also known as Megaco Defined as IETF RFC 2885. Megaco defines a centralized architecture 9
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 10
H.323 Components GK H.323 MCU Scope of H.323 e H.323 Gatekeeper Packet Network H.323 Terminal H.323 Gateway PSTN ISDN V.70 Terminal H.324 Terminal Speech Terminal H.320 Terminal Speech Terminal 11
Scope of H.323 Recommendation Video I/O Equipment Audio I/O Equipment Video Codec H.261, H.263 Audio Codec G.711, G.722, G.723, G.728, G.729 Receive Pain Delay (Sync) UDP RTP RTCP User Data Applications T.120, etc. System Control H.225 Layer IP H.245 Control TCP System Control User Interface Call Control H.225.0 RAS Control H.225.0 UDP 12
H.323 Signaling V H.323 Endpoint A Setup Alerting / Connect V H.323 Endpoint B H.225 (TCP Port 1720) Capabilities Exchange / MSD Open Logical Channel Open Logical Channel Acknowledge H.245 (TCP Dynamic Port) RTP Stream RTP Stream RTCP Stream Media (UDP) 13
Basic H.323 Call Gatekeeper A ACF LRQ LCF Gatekeeper B ACF RRQ/RCF IP Network RRQ/RCF ARQ V Gateway A H.225 (Q.931) Setup H.225 (Q.931) Alert and Connect H.245 RTP V ARQ Gateway B Phone A Phone B 14
Deploying H.323 Networks DGK Minimizes GK configuration Addition of new zones Addition of new NPAs Addition of new rate centers GK GK GK LA West Zone LA GW #1 GW #2 Rate Intra-LATA Rate Center #1 Toll Center #1 Chicago GW Local Midwest PSTN Zone NY GW Local East PSTN Zone 15
MGCP/H.248/Megaco Architectures Call Agent Call Agent SS7 P S T N IMT PRI PSTN P S T N Access Gateway MGCP / H.248 / Megaco RTP 16
Deploying MGCP/H.248/Megaco Networks STP PSTN SS7 SS7 Backhaul SLT IMTs MG CA Traditional TDM Traffic Modem Dial-up Traffic MGCP and ISDN Backhaul MG VoIP ISDN/PRI OSS TDM Voice NAS/VoIP Billing and Measurement Server Service Provider's TDM Network Service Provider's Packet Network 17
SIP Architecture I N T E L L I G E N T email LDAP Oracle XML SIP Proxy, Registrar & Redirect Servers SIP CPL 3pcc Application Services S E R V I C E S SIP SIP SIP User Agents (UA) RTP (Media) PSTN Legacy PBX CAS or PRI 18
SIP Signaling PSTN Calling Party SIP Signaling and SDP Signaling (UDP or TCP) SIP VoIP Network INVITE 100 Trying INVITE 100 Trying 180 Ringing 180 Ringing 200 OK 200 OK PSTN Signaling Called Party ACK ACK Media (UDP) RTCP Stream Bearer Or Media 19
SIP Servers/Services Registrar Redirect Location Database SIP Servers/ Services Where is this name/phone#? REGISTER Here I am 3xx Redirection They moved, try this address INVITE I want to talk to another UA SIP Proxy Proxied INVITE I ll handle it for you SIP User Agents SIP User Agents SIP-GW 20
Deploying SIP Networks PSTN 312 Chicago POP Central Zone PSTN 212 NY POP East Zone IP Network West Zone SF POP PSTN 415 21
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 22
Voice Myths Myths Networks can only be built one way Networks will only use one protocol All networks will converge Facts VoIP allows centralized or distributed architectures H.323, SIP, MGCP and H.248/Megaco will all be present in VoIP networks Networks will converge to IP 23
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 24
Interconnecting VoIP Networks H.323 SIP? MGCP H.248 Megaco 25
Connecting VoIP to SS7/C7 Networks IAM H.323 H.225 Setup (ANI,DN) Proceeding H.245 MGCP CRCX ACK SDP SIP INVITE ACK SDP 26
VoIP Interworking Issues Service interworking E.g.: H.450 <-> SIP <-> MGCP Media interworking End-to-end codec negotiation Bearer interworking End-to-end fax, modem, DTMF 27
VoIP Interworking Bearer level Modem (relay/passthru) Fax (relay/passthru) T.38 T.37 DTMF (relay/passthru) Media level Codec (negotiation, selection) Service translation issues Call deflection Park/hold Signal issues SDP H.245 28
Fax and Modem Passthru Mechanisms Modem and fax are control mechanisms based on PLL (Phase Locked Loops) They are both time sensitive Highly sensitive to packet network impairments: Jitter Packet loss Delay Susceptible to clock slew (clock sync differences between gateways) 29
Passthru Simplified Voice Gateway PCM G.711m DSP G.711m G.729 IP Cloud Voice Gateway G.711m G.729 DSP PCM G.711m 30
What Is Modem Passthru? It is the transport of modem signals (modulation, error correction and compression) through a packet network using PCM encoded packets 31
Modem Passthru (Cont.) Modem tone detection (<= V.90) Switchover signaling No VAD EC off RTP payload redundancy (10ms packetization) RFC2198 (optional) 32
Modem Passthru Issues Consecutive packet drops (loss) cause retrain Consecutive drops during retrain causes disconnect Variation of delay (jitter) has quite an effect Jitter (at 10%) is a conservative estimate Since jitter mostly impacts performance with packet loss 33
What Is Modem Relay? Modem relay involves demodulating the modem signal at ingress gateway Passing this data as packet data to terminating gateway Re-modulating the data and passes it to the receiving modem 34
Fax Relay T.38 T.30 UDP T.30 PSTN PSTN Real-time Also called demod/remod Can be used in H.323/MGCP/SIP signaling Delivers fax data over UDP streams (uses same RTP port) reuses voice UDP ports Fallback to proprietary mode Method of encoding the T.30 and T.4 into packets IP 35
DTMF What is DTMF Why is it required? and where is it used? How do you transport it in IP? DTMF implementation 36
DTMF (Cont.) In TDM world, all voice traffic is sent as uncompressed 64Kbs PCM streams; anything sent on that circuit is an untouched stream of bits; (e.g., voice speech, modem tones, fax tones, and DTMF digits) DSP codecs designed to interpret human speech, can distort DTMF tones (machine-tones) High b/w codecs less likely to distort Distortion causes problems with voicemail and IVR systems 37
DTMF Schemes with VoIP Protocols H.323 MGCP, H.248, Megaco SIP In-Band In-Band In-Band In-Band Out-of- Band Cisco RTP, H.245 Alphanum, H.245 Signal, AVT Tones RFC2833 Cisco RTP, NSE, NTE,RFC2833 RFC2833 38
Topics to Discuss History of VoIP VoIP Early Adopters VoIP Standards and Standards Bodies VoIP Making Sense of the Protocols The Great Voice Myth VoIP Protocol Challenges Summary 39
Summary Understand the possibilities and the issues Avoid protocol/product based bias Decide on application Consider market and business drivers Deploy what s possible today Choose signaling protocol depending on services intended to be offered Many possibilities stay tuned 40
Crystal Ball on VoIP All three protocols (or its variations) are here for the long run Changes/enhancements will be made IP will be the core 41
Reference URLs ITU: IETF: SIP: www.itu.org www.ietf.org www.cs.columbia.edu/~hgs/sip/ H.323: www.packetizer.com/iptel/h323/ MGCP:www.softswitch.org/asp/techlibrary _protocol.asp?page=techlibrary 42
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