VoIP FundamentalsInternet2 TechEx

Workshop

VoIP Internet2 Technology Evaluation Center (ITEC)Walt Magnussen Ph.D. TAMU

Jason McConnell TAMUBen Fineman Internet2

John Hird Mitel15 October, 2017

Schedule• 8:00 am - Introductions• 8:15 am SIP/VoIP fundamentals – Walt Magnussen• 10:00 am Break• 10:15 am Communications Directions – Ben Fineman• 10:30 am Hands on workshop – Jason McConnell

– SBC– Gateway– Devices

• 11:30 Op-Easy John Hird• 11:50 Walt Magnussen

VoIP Fundamentals

The TechnologyThe IndustryRegulation

Telephony basics

• In today’s world all voice is converted to digital

• A/D conversion can be done in phone or in or switch

• Traditional– Circuit Switch– Packet based

Traditional Telephony TDM

PSTN

Analog

AnalogDigital

Digital

CircuitEstablished

VoIP TelephonyPSTN

ILEC and/orIXC

Internet orPrivate

Network

VoIPPhone

VoIPPhone

CallManager

Router

Router

EthernetSwitch

EthernetSwitch

`WirelessAccess Point

EthernetSwitch

Gateway

Call DataOnly

SignalingOnly

Encoding and VoIP packets

Steps1.) Sample – 8,000 samples per second2.) Quantify – Each sample is 8 bits – S2=111110103.) Encoded to create data stream4.) In the case of VoIP data put into packets

S1 S2…….Sn

Analog Digital

11111111

11110000

00000000

Date Packet 1 Data Packet 211001101010101001001010010010010101001010010101010

Ethernet Data Packet 1 SIP Header IP Header Ethernet HeaderTrailer

VoIP Data Packet

TDM vs. VoIP

• TDM– Circuit switched -Dedicated Pipe

– Advantage – no congestion issues– Disadvantage- inefficient

• VoIP– Packet Switch - Shared path

– Advantage – no need to build separate network– Disadvantage – needs QoS, large pipes or great luck

Signaling Protocols• H.323

– Cisco SKINNY• MGCP

– Was used extensively by carriers• Modified SIP - Lync• Then there was SIP (Session Initiated

Protocol)– Has won the standards war

What’s SIP• IETF RFC 3261

– Replaces RFC 2543• “The Session Initiation Protocol (SIP) is an application-layer

control (signaling) protocol for creating, modifying and terminating sessions with one or more participants.”

• Can be used for voice, video, instant messaging, gaming, etc., etc., etc.

• Follows on HTTP– Text based messaging– URIs – ex: sip:cnorton@TAMU.EDU

Where’s SIP

Application

Transport

Network

Physical/Data Link

Ethernet

IP

TCP UDP

RTSP SIP

SDP codecs

RTP DNS(SRV)

SIP Components• User Agents (UA)

– Clients – Make requests

– Servers – Receive requests

• Server types– Redirect Server

– Proxy Server

– Registrar Server

– Location Server

• Gateway– UA connecting to

another network – eg. the PSTN

• B2BUAs– Two UAs that pass

SIP messages – and can modify them

SIP TrapezoidDNS

ServerLocation Server

Terminating User Agent

Outgoing Proxy

Originating User Agent

DNS

SIP

SIP

SIP SIP

RTP

Registrar

Incoming Proxy

SIP

SIP TriangleDNS

ServerLocation Server

Terminating User Agent

Originating User Agent

DNS

SIP

SIP SIP

RTP

Registrar

Incoming Proxy

SIP

SIP Peer to Peer!

Terminating User Agent

Originating User Agent

SIP

RTP

SIP Flows - Basic

ACK

200 - OK

INVITE: sip:18.18.2.4“Calls” 18.18.2.4

180 - Ringing Rings

200 - OK Answers

BYEHangs up

RTPTalking Talking

User A

User B

SIP StandardsJust a sampling of IETF standards work…

IETF RFCs http://ietf.org/rfc.html

• RFC3261 Core SIP specification – obsoletes RFC2543

• RFC2327 SDP – Session Description Protocol

• RFC1889 RTP - Real-time Transport Protocol

• RFC2326 RTSP - Real-Time Streaming Protocol

• RFC3262 SIP PRACK method – reliability for 1XX messages

• RFC3263 Locating SIP servers – SRV and NAPTR

• RFC3264 Offer/answer model for SDP use with SIP

SIP Standards (cont.)• RFC3265 SIP event notification – SUBSCRIBE and NOTIFY

• RFC3266 IPv6 support in SDP

• RFC3311 SIP UPDATE method – eg. changing media

• RFC3325 Asserted identity in trusted networks

• RFC3361 Locating outbound SIP proxy with DHCP

• RFC3428 SIP extensions for Instant Messaging

• RFC3515 SIP REFER method – eg. call transfer

• SIMPLE IM/Presence - http://ietf.org/ids.by.wg/simple.html

• SIP authenticated identity management -

http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-06.txt

Infrastructure Requirements

• Converged data network is the underlying data infrastructure.– VoIP killers (packet loss, jitter and latency)

• ITU-T G.1050 Specification– Well Mangaged – Strict QoS No oversubscription -

High quality VoIP and video– Partially Managed – Separate queue with preferential

treatment – VoIP and VTC– Unmanaged – Low quality VoIP and VTC, Signaling

transactions

Packet Jitter cont’d

• A jitter buffer is a queue in the phones that receives the arriving packets and sends the packets out in a equally spaced time interval.

Example of packet due to Jitter

Packet loss impactPingtel MOS (PESQ LQ)

0.751.001.251.501.752.002.252.502.753.003.253.503.754.004.254.50

0% 5% 10% 15% 20% 25% 30% 35% 40% 45% 50%Packet Loss

MO

S (P

ESQ

LQ

)

G.711GIPS G.711G.729

ITU-T G.1050 specification

• Impairment Type Units Range (min to max)– Latency

• One Way latency ms 20 to 100 (regional)• 90 to 300 (intercontinental)

– Jitter (peak to peak) ms 0 to 50– Random Packet Loss % 0 to 0.05– Reordered Packets % 0 to 0.001

Infrastructure Strategies

• Separate Network (Physical separation)• Over Provisioning• Logical Partitioning

– VLANS (802.1q) for local area network– MPLS for wide area networks

• Prioritization– Layer 2 (802.1p)– Layer 3 IP - ToS

Powering devices• Most all ethernet switching manufactures

now support POE.• Standards well defined IEEE 802.3af

Power over Ethernet (POE) Standards • Several Pre-standard

implementations• June 2003 IEEE ratifies

802.3af standard– PSE or Power Sourcing

Equipment– Powered Device – VoIP

phone, AP or camera• PSE senses power

requirement before it applies power

• Good white paper http://www.panduit.com/enabling_technologies/098749.asp

Cost of supporting QoS

• Multiple VLAN support• Mapping application or device to VLAN• Mapping VLANs to MPLS tags• Marking Priorities

– At device– At edge of network

• Mapping priorities for end to end (802.1p to ToS)

Common Solutions for Universities• Carrier Hosted Solutions• Vendor CPE Solutions• Open Source Solutions

Carrier Hosted Solutions

• Verizon, AT&T and Century Link Hosted –require IP access service as well

• Other parties, Vonage, Packet8 etc. Over the top

• Internet2 SIP Cloud Service – on Internet2 network

Verizon HIPC

• Uses Broadsoft’s Broadworks Platform• Designed to be a carrier class VoIP solution also

known as Hosted IP Centrex (HIPC)• Distributed platform design containing three

major components Application, Network, and Media Servers

• Each 3 component node can be expanded to be physically or geographically redundant

CPE Solutions

• AASTRA Clearspan – Same as Internet2 hosted (merged with Mitel)

• Broadsoft • Cisco Unified Communication Manager

(CallManager)• Microsoft Lync• Avaya• Others

Cisco Call Manager• Designed to be an

enterprise based VoIP solution

• Single server platform that runs on a x86 platform with Microsoft Windows or Linux as the operating system

• Platform can be replicated for redundancy

Cisco Call Manager

• Supports only Cisco SKINNY and SIP protocol on the IP phone side

• Supports SKINNY, H.323, and SIP on IP trunking side

• Ability to integrate with approved 3rd party applications for Voicemail and other features

• Supports up to 2,500 users per server with clustering user count can scale to 10,000

Open Source Solutions

• IPTel SIP Express Router (SER)• Asterisk• Not implimented as long term solution.

Asterisk• Designed to be a solution for small to medium business• Can be scaled for larger campuses• Runs on most any Linux or Unix based x86 platform

including OS X• Provides many PBX like features including voicemail and

music on hold• System supports SIP, H.323, IAX, SKINNY, and MGCP• External gateway or PC telephony card for PSTN

connection.• No formal support structure for system. Community

support forums used to troubleshoot issues.

IP Trunking

• Local IP trunks• LD IP trunks• IP peering

Local IP trunks

• Typically CLEC offering– Level 3– Verizon– AT&T– Century Link

• Used by local VoIP service providers (Vonage)• Supports LNP• Offered by many service providers.

LD trunks

• Terminates LD traffic on net• Exlusively SIP offering• Many services provided

– Toll call– Inbound and outbound 800– Directory Assistance

• Eliminates need for PRIs (1.7 Mbps per 23 trunks when using SIP)

Selecting the Right Telephone

• Desk (hard) phones– Multi-line IP phones– Speaker phones– Supporting analog devices

• Mobile Phones– Soft Phones– Wi-fi Phones– Dual Mode Phones

Selecting Protocol

• Skinny – Cisco - Several Hundred features• Lync – Microsoft limited feature set• SIP Proxy – AASTRA Clearspan

(Broadsoft), Mitel, Avaya. Full feature set.• SIP instruments – Polycom, Mitel, Cisco

plus several others

Multi line instruments Cisco

Multi Line InstrumentsPolycom

Supporting Single Line Analog Devices

Soft Phones

• Typically installed on a laptop for personnel on the go.

• Requires a Wi-Fi or Wireless connection and a headset.

• Does not require an additional instrument to place the call.

• Soft phone has to compete with other applications running on laptop

Soft Phone Examples

Cisco IP Communicator(SKINNY)

CounterPathEyeBeem

(SIP)

KPhone (SIP)

Wi-Fi Phones

• Dedicated instrument to place the call.• Requires Wi-Fi available location.• Issues with special authentication

measures at some hotspots.• Battery life still an issue, but getting better.• Call will terminate when Wi-Fi signal too

low.

Wi-Fi Phone Examples

Zytel P2000W (SIP)

Cisco 7920 (SKINNY)

Linksys WIP300 (SIP)

UTStarcom F1000B (SIP)

Dual-mode Phones

• The ultimate goal in mobility.• Biggest issue roaming between networks.• Battery life major concern.• Dedicated instrument.

Dual-mode Phone Examples

IPhone 8

Galaxy S8

IPhone 8

Nokia 8

VoIP Peripherals

• Conference Bridge• Voicemail with Unified Messaging• Auto-attendant• ACD

Implementation Issues

VoIP Security

• Network protection– Firewall – difficult to map signaling to media

stream (stateful required)– VLAN isolation– Session Border Controller (SBC) back to back

user agent.• SPIT• Sequential dialing• VoIP DoS or fuzzing (deep packet inspection)

Authentication

• Authentication– Skinny– SIP

• Encryption– TLS (SSL v3) signaling– SRTP or SRTCP media

E911

• Requirement is a State by State decision (i.e. Texas requires if University has residence halls).

• Is complicated by VoIP but still supportable.

CarrierCentral Office

Selective Router

PSAPALI Database

Campus VoIPProxies

CAMA Trunk or ISDN PRICampus Telemanagement Server

PSALI Service Provider

PSAL

I Upd

ate

MSAG validated ALI update

911 Call

E911 Architecture

VoIP 911 Issues

• Fixed Phones – Mobility– Phones can be nailed to an ethernet port by

locking MAC address to port• Nomadic phones – Soft and Wireless

phones– Tag as nomadic in ALI database

VoIP Checklist• Organizational

– Relationship between Voice and data• Document roles for each side if separate• Help desk

• Infrastructure– Data Network assessment

• Capabilities of switches• VLAN or QoS support• POE • Battery / backup power for all critical devices

VoIP Checklist• Server Selection

– Signalling protocol – SIP vs H.323– Hosted– Private Servers– Open Source

• Feature Set requirements– Select instruments

VoIP Checklist• Select Implementation strategy

– Cost of supporting dual systems– Migrating trunks and number pools from one

switch to another– Select trial group– Establish rollout plan

VoIP Checklist• Managing customer expectations

– It will be different!– Diagnostics can and are more complex

• Training– Customer Service Reps– Technicians

• Traditional data techs manage Infrastructure• Traditional voice techs now manage application

that runs on network

VoIP Checklist• Network connections

– Gateway placement and type– IP trunks or PRI trunks

• E911– Lock ports or allow mobility (802.1x)

• Security– SBC, VoIP aware Firewall or Open Network

VoIP Checklist• Funding Models

– TDM easier to calculate– VoIP

• Proxy cost• Instrument costs• Infrastructure – POE switch ports, Cat 5E or 6

cable, Backup power, QoS management.

Deeper dive of architecture

10/19/1762

IMS for Higher Education –Conceptual view• Desired Services would include;

– Find me-follow me roaming between Enterprise Voice and VZW

– Presence updates using SIP notify messages– Enterprise phone number used as CID– Support for IMS App Server from VZW network– Enterprise Voice Mail integration– Internet2 network supported as a Visited Network

with full policy support including QoS.– WiFi voice offload

The high level view

Resources Available• Core SIP proxy – Broadsoft Broadworks• IMS Core – ACME Packet • LTE

– Motorola Ericsson Core and RAN– General Dynamics LTE EPC core– Juniper MX router with AMT– OctoShape Video

• NG 9-1-1– US DoT POC system– Geocomm LoST– Avaya ESRP– Redsky LIS– Acme Packet BCF

ENUM Call routing• Campus to campus direct SIP calling• ENUM is based upon RFC 3761• Widely used in Asia and Europe• Internet2 has obtained the root for the US.

– 1.nrenum.net– Example for TAMU

• 5.4.8.9.7.9.1.nrenum.net

• Routing by SBC in our case.

NG 9-1-1• Efforts begin by NENA in 2004• i3 standards developed• Interop testing in progress• Tests and early deployment in several

states.

NG 9-1-1 architecture

What we can do• Work on LIS• Follow local PSAP status• Enable local VoIP SIP server or proxy to

make NG 9-1-1 calls.

ENUM Enabling Collaboration

Walt Magnussen, Ph.DDirector ITEC Texas A&M

University14 January, 2014

Problem Question• How do you keep on-net traffic on-net?• How do you discover the best way to route a call

to a collaborator?• How do you make real time voice and video

networks one?• How do you conference between non-

interoperable video networks?– H.323– SIP– Telepresence

Answer• Use SIP as the common signaling platform• Enable DNS based call lookup• Gather critical mass in higher education

ENUM – What is it• Mapping E.164

telephone number to URI using DNS

• Operated and Governed by NRENs

• Recognized by RIPE ENUM WG (e164.arpa)

• IETF RFC 4769 tells you how to impliment

NRENum.net• 30 Country Codes register (5 of them in

e.164 registry)• Approximately 184,000 numbers

registered• Internet has been delegated +1 and is

running the registrar

High Level ENUM

ENUM call flow

Least Cost Routing• Layering of Call routes

– Local – ENUM– Arbitrage (i.e. InteliPeer)– LD Service provider

Creating SRV records• 8.8.9.4.8.5.4.9.7.9.1,

9794584988@voip.tamu.edu

• 0.6.1.9.8.5.4.9.7.9.1, 9794589160@voip.tamu.edu

• 5.6.4.0.8.5.4.9.7.9.1, 9794580465@voip.tamu.edu

This is more than just VoIP• Bridging

– SIP VoIP E.164– H.323 GDS 01-751-55678-99215– Telepresence 1-751-555-1234

Bringing ENUM to your campus• Will support any platform

– Internet2 Net Plus– Enterprise VoIP (Cisco, Avaya, Genband etc.)– TDM with the addition of an SBC

• Get copy of ENUM cookbook• Attend Internet2 collaboration workshop

(Denver meeting in April)• Contact us for help

IMS for Higher Education• SIP in the Cloud service went on line in

November 2012• ITEC Advisory Committee met on Monday

for the first time• Asked to work on new services including

– FMC over IMS– ENUM for Higher Ed– NG 9-1-1 native from Proxy– Lync Integration

IMS for Higher Education –Conceptual view• Desired Services would include;

– Find me-follow me roaming between Enterprise Voice and VZW

– Presence updates using SIP notify messages– Enterprise phone number used as CID– Support for IMS App Server from VZW network– Enterprise Voice Mail integration– Internet2 network supported as a Visited Network

with full policy support including QoS.– WiFi voice offload

Questions• Thank You