Troubleshooting SIP

5 Day Course
Hands On
Code PDN024

Book Now - 3 Delivery Methods Available:

Scheduled Online Onsite

Overview

Next Generation voice services are being delivered using SIP signalling. SIP based Voice over IP promises to reduce telephony costs and provides unique opportunities for integrating voice and Internet services. Voice services are being provided from SIP based telephones, soft phones and phone adaptors from customer premises over DSL access. SIP servers located within telecommunications service providers can deliver connectivity to external and international carriers but the telecom authority may not be the authority which deliver Internet services directly to the customer. In the event of failures in the services, technical staff need to be able to troubleshoot the voice services to discover the source of the fault locating it to customer premises equipment failure, configuration failures in customer equipment, failures within ISP service equipment or failures within the carrier voice services. Also increasingly customers are themselves implementing SIP based PBXs on their own site and wish to carry trunk connections using SIP between sites or into carrier services.

This course will provide an understanding of how to locate and troubleshoot faults within SIP based operational VoIP services. It will assume that the staff attending have already undergone initial training in the operation of the SIP signalling protocol for delivery of VoIP. It will, through practical hands-on workshops teach the SIP troubleshooting skills needed to recognise faults, locate them to the failing component and to fix these where that is feasible. The course will also provide an understanding of how SIP is being used between SIP based PBXs on customer sites acting as trunking paths to other SIP servers, both within the customer premises and within the carrier environment. It will also teach how this SIP trunking can be used for inter-carrier connection in the future. As ISDN based voice services declines, the voice features for call transfer, diversion, and other service features must be provided using SIP. The course will demonstrate how these key services may now be delivered using SIP and how to troubleshoot failures.

The course will use PC based protocol analysis software to isolate signalling faults and to reconstruct voice carried over the network in order to monitor voice quality. The course will address the different mechanisms used within IP and Ethernet protocols to deliver QoS and how QoS is affected by failures caused by incorrect sizing and network design at both the customer site and within the IP network delivered by the ISPs.

Objectives

When you have completed this course you will be able to:

  • Describe the way SIP is used for delivery of Voice services over DSL access, over SIP trunks and to deliver extended voice services
  • Isolate and troubleshoot faults and failures in SIP based VoIP Services
  • Analyse the key VoIP protocols used to deliver quality voice and troubleshoot quality failures
  • Use PC based protocol analysers to reconstruct and record voice channels to monitor voice quality
  • Troubleshoot IP reachability problems associated with third party ISP access over DSL using PPPoE
  • Size a VoIP service for both enterprise and carrier environments
  • Monitor trunking channels through Cisco routers and switches
  • Troubleshoot SIP trunking interfaces between SIP services over both CPE and carrier SIP based switches

Modules

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SIP Architecture Review (4 topics)

  • Where SIP fits into Next Generation Voice services
  • SIP Phones and Adaptors
  • SIP Service Architectures
  • Review of Simple SIP Signalling Exchange

Hands On Session 1: Here is SIP Voice Over IP Working (3 topics)

  • Set up and use SIP VoIP applications on each PC to place calls across the classroom
  • Setup SIP phones/phone adaptors
  • Place and receive SIP calls within the classroom

Elements of Troubleshooting (15 topics)

  • Purpose and goals of troubleshooting
  • Stages in troubleshooting
  • Defining a working service
  • Identifying the symptoms
  • Defining the problem
  • Speculating upon causes
  • Proposing tests to validate causes
  • Testing to discover cause
  • Fixing the problem
  • Validating the solution
  • Testing tools
  • Observation
  • Call connect testing
  • Inbuilt testing services
  • Protocol analysis

Hands-on Session 2 Elementary test Capture (3 topics)

  • Observing SIP Phone/Adaptor configurations through browser
  • Set up and use Protocol analyser to capture SIP VoIP traffic
  • Observing working SIP calls

Elementary Failure modes (13 topics)

  • Normal SIP exchanges
  • Registration
  • SIP URIs
  • Authentication
  • Call initiation
  • Carriage of ringing and call progress
  • Call connection
  • Establishing voice channels
  • Clear-down
  • Session Descriptions
  • Media streams
  • Attributes
  • DTMF tone options

Hands-on Session 3 Analyzing SIP Exchanges (3 topics)

  • Capturing and Analyzing call exchange
  • Identifying configuration errors
  • Analyzing codec selection

Network Failure Modes (7 topics)

  • Reachability issues
  • IP address reachability testing
  • Port reachability
  • Problems with firewalls
  • Network Address Translation Issues
  • Port translators
  • STUN and other solutions for NAT traversal

Hands-on Session 4: Observing and troubleshooting connection issues (3 topics)

  • Analysis of services to identify address and port failures
  • Observing NAT traversal problems
  • Fixing port reachability through firewalls

Internet Service Failures (8 topics)

  • Addressing and routing issues
  • Identity duplication
  • Typical routing mistakes to look for
  • Impact of packet loss
  • Known SIP weaknesses
  • Impact of variation in delay on voice channels
  • RTP and RTCP time and clocking issues
  • NTP

Hands-on Session 5: Analyzing Internet failures (3 topics)

  • Capturing and Analyzing call exchanges under packet loss conditions
  • Reconstructing voice channels to identify voice quality
  • Observing timing issues

SIP Proxy Servers and Trunking (7 topics)

  • Functions of SIP Proxy
  • SIP Realms
  • Calls between different Realms
  • SIP Trunks
  • Example SIP Trunk configurations
  • SIP URI allocation
  • Deploying Numbering plans

Hands-on Session 6 Configuring SIP Trunks (3 topics)

  • Configure SIP trunk between different SIP Servers
  • Connecting calls between Proxy Servers
  • Observing SIP Trunk Signalling

Troubleshooting SIP Trunks (5 topics)

  • Identity and authentication
  • Call connection signalling
  • Port allocation
  • Race conditions and avoiding them
  • Sizing and loading issues

Hands-on Session 7: Troubleshooting SIP Trunk connection (3 topics)

  • Identifying SIP Trunk Failure conditions
  • Measuring call loads
  • Filtering out Non-SIP traffic

Troubleshooting SIP Supplementary Services (5 topics)

  • Call Transfer
  • Call Diversion
  • Diversion on Busy
  • Diversion on No Answer
  • Voice Mail

Hands-on Session 8 :Troubleshooting Supplementary Services (3 topics)

  • Analyzing SIP Diversion Traffic
  • Analyzing SIP Transfer Traffic
  • Identifying key failure conditions

QoS in a Multi-supplier Service (8 topics)

  • Troubleshooting in a mutli-vendor environment
  • Identifying points of demarcation
  • Validating voice quality issues
  • Quality of service issues
  • Sizing and queuing
  • 802.1p priority
  • DiffServ in IP
  • VLANs using 802.1q

Hands-on Session 9: QoS (3 topics)

  • Identifying voice quality in a multi-vendor network
  • Observing QoS issues
  • Implementing router port monitoring

Hands-on Session 10: Putting it all Together (2 topics)

  • Practical troubleshooting of service configuration to find and correct faults
  • Each student team will configure a different SIP service with a fault and the other teams will take turns at isolating the faults.

Prerequisites

The course is aimed at technicians and engineers who have already undertaken training in TCP/IP and elementary SIP VoIP. It assumes a working knowledge of Microsoft Windows and simple TCP/IP networking. Elementary telecommunications knowledge is also assumed and the ability to configure SIP phones/adaptors. Some experience of using Cisco routers or other router configuration would be an advantage.

Additional Learning

The courses below may help you meet the knowledge level required to take this course. If you are unsure please ask a training advisor .

Scheduled Dates

Please select from the dates below to make an enquiry or booking.

Pricing

Different pricing structures are available including special offers. These include early bird, late availability, multi-place, corporate volume and self-funding rates. Please arrange a discussion with a training advisor to discover your most cost effective option.

Code Location Duration Price Dec Jan Feb Mar Apr May
PDN024

(Classroom or Online)
5 Days $2,995

What Our Customers Say

The training was delivered with a high level of expertise and excellence. Instructor was highly knowledgeable.”

Technical Trainer, Aviat Networks

Overall the course was really good, the trainer really understood the material and was very approachable.”

Customer Training Manager, Aviat Networks

Excellent course, informative and well-paced.”

CSE, Cisco

Course was very well outlined. Topics were great and bridged many gaps.”

System Engineer, Cable & Wireless

An excellent intro to video encoding & MPEG transport streams - I would definitely recommend it.”

Broadcast Engineer, Cisco

Definitely an excellent intro. Left me interested in learning more.”

Broadcast Engineer, Eircom

Excellent training course with real examples and practical classroom demonstrations.”

Transport Designer, Orange

Instructor knowledge and experience was excellent.”

Solutions Engineer, Akamai

Excellent course, very clear and well organised. Course content delivery was very good.”

Assistant Engineer, Dhiraagu

Very informative and appropriate.”

Network Support Technician, BT

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