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:[email protected]
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,
• 0.6.1.9.8.5.4.9.7.9.1, [email protected]
• 5.6.4.0.8.5.4.9.7.9.1, [email protected]
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