Sangoma SBC Training Presentation

Sangoma Session Border Controllers:

Support Training Presentation

Aykut [email protected] Turkey Distributor

Contents Part 1• Why Session Border Controllers?• Product Portfolio of the Session Border Controller• Business Applications and Use Cases (Vega ESBC)• Carrier/Service Provider Applications and Use Cases

(NetBorder SBC)• Sangoma SBC Load Balancing and Failover Techniques• SBC Walkthrough

© 2014 Sangoma Technologies 2

Contents Part 2• Conceptual Overview of the SBC Call Processing Components• Introduction and Configuration of SIP Profiles• Introduction and Configuration of Domain Profiles• Introduction and Configuration of Media Profiles• Introduction and Configuration of SIP Trunks• Introduction and Configuration of Call Routing• Walkthrough• Questions


Reasons for Session Border Controllers

Problems Found in VoIP without an SBC

• Firewalls need to be traversed for end-to-end VoIP telephony– SIP protocol does not work through NAT functions in firewalls

• Without SBCs– Forward SIP/RTP ports on firewalls

• Opens up security issues

– Set-up VPNs• Costly to manage/bandwidth limitations/subscriber mgmt.

– Firewall Application Layer Gateways (ALG)• OK, brings other limitations for other SIP issues


SIP Interoperability Challenges

• SIP RFC3261– Largest RFC– Not a tight specification like ITU specs for instance– Uses “Should”,“Can”,“May”,“Option” many many

times– It is a recommendation, not a hard rule, lots of

room for interpretation

• Result– Everyone is compliant to RFC3261– But hard time to interop!

• For end to end VoIP Interworking, SBCs come to the rescue by ‘fixing’ these differences


Additional Interop Challenges• It’s not just SIP signaling• Media can also need fixing for end to end communications to become

possible:– Codecs mismatch– Fax T.38/Inband Fax– RFC2833/INFO/Inband DTMF Methods– RTP and SRTP

• IPV6 vs IPV4• UDP vs. TCP (example with MS Lync)• TLS/SRTP interop with SIP/RTP• Firewalls cannot address these – do not have DSPs to process media


Security Issues• Connectivity to other IP Networks introduces

security issues– Denial of Service (DoS) attacks– Toll Fraud by manipulating media– Topology hiding (SIP vias, hops, etc.)

• Firewalls cannot act on all these security issues unless it is SIP aware (SIP ALG)– Some firewall vendors offer SIP ALGs, but it is not enough


What is an SBC and Why have one?• SBCs are installed at the edge of VoIP Networks to facilitate

end to end VoIP transmission without compromising network security

• Essential for several reasons:– New security issues introduced with SIP protocol– Fix interoperability issues

• SBC are typically implemented as Back to Back User Agents (B2BUA)– All SIP and media (voice) traffic transit through SBCs


B2BUA Explained• A back to back user agent (B2BUA) is a logical network

element in the Session Initiation Protocol (SIP) applications

• It operates between two endpoints in a communication session and divides the communication channel into two different call legs

• It mediates SIP signaling between both ends of the call• B2BUAs are often implemented within media gateways


B2BUA Explained


• SIP Normalization• Security• Transcoding• CDRs• RTCP QoS report• Call Access

Control• Management• GUI / config• DSP resources• Etc.

Eth pipeSIPport

RTP ports

SIPport

RTP ports

Eth pipeSIP

Media

SIP

Media

SBC

Because the SBC ‘sees’ all SIP and RTP traffic coming from both sides, it can analyze, fix, control, etc.

• SIP Normalization• Security• Transcoding• CDRs• RTCP QoS report• Call Access

Control• Management• GUI / config• DSP resources• Etc.

Eth pipeSIPport

RTP ports

SIPport

RTP ports

Eth pipeSIP

Media

SIP

Media

SBC

Where are the User Agents (UA)?


SIP UA SIP UA

They are back to back!

Product Portfolio of theSession Border Controller

Vega Enterprise SBC• Appliance

– 25-250 Sessions– H/W DSP acceleration– 1U/2 x 1 GE ports

• Software Version– 25-500 Sessions/Self-Contained ISO– VM requirements

• 1 Core/1 GB RAM/Bridged

• Software/Hybrid Version – UNIQUE– 25-500 Sessions/Self-Contained ISO– VM requirements


– H/W DSP acceleration


D150

NetBorder Carrier SBC

• Appliance– 250-4000 Sessions– H/W DSP acceleration– 1U/2 x 1 GE ports– RAID 1



Product Highlights – All SBCs• Ease of Use

– WebGUI configuration, operation, backup and restore, REST API– Simplified licensing, field upgradable, all features one SKU

• Session Policy and Media– Advanced WebGUI or XML header manipulation, upper registration– NAT traversal, call forking

• Security– DDOS attack protection, advanced firewall for signaling and data

• Advanced Call Routing– Advanced WebGUI or XML dialplan, database routing, load balancing

• Troubleshooting– PCAP signaling and media capture on the SBC, email notifications

• Redundancy/HA– Active - Active or Active - Standby

Ease of Use• WebGUI configuration, operation, backup and restore

– Golden master configuration used to configure multiple SBCs

• REST API– Integrate Sangoma SBC into a business process – Programmatic SIP trunk and user configuration

• Simplified licensing, field upgradable– All features one SKU. No feature limitations. – Transcoding, SRTP, voice quality features all included– Sessions are software upgradable from 20 to 250 sessions

• Email Notifications– Notifications on error conditions, failures, security or capacity


Session Policy• Advanced Header Manipulation

– GUI or XML based manipulation of any SIP header on any SIP packet. INVITE, 180,183,200, etc…

• Upper Registration – Remote Users– Pass-through registration with in and/or out of dialog support– Advanced call flow scenarios to support remote users

• NAT Traversal– Auto IP detection

• Call Forking– Multiple outgoing dialogs per call

• First 200 Ok receives the call, rest of the calls get hung-up• Support for busy, unregistered or inactive user agents

• Unlimited SIP Interfaces• Unlimited SIP Trunking• SIP and Media Transports

– TCP, UDP, TLS, RTP, SRTP


Media and Networking• Hardware Media Processing

– Sangoma SBCs use hardware network DSPs to process RTP– Low latency media pass through– High capacity any to any transcoding and encryption– Voice quality enhancements

• Echo cancellation, noise reduction, auto gain control

• Networking– Single IP address for signaling and media– Separated signaling and media planes – VLAN and ethernet bonding


Security• Signaling Security

– Adaptive and time based firewall blocking based on SIP flood attacks • Malformed packet, registration storms, invite floods, authentication errors

– SIP scanner detection and blocking– Rule based detection and blocking

• Using standards based rules and known exploits and blacklists

• Media Security– RTP media port pin hole based on active session

• RTP ports are only opened when session is active

– RTP port overload detection. In case of RTP flood attack on a specific port.

• Data Firewall– Advanced state full data firewall – Port forwarding and NAT

• DDOS– Adaptive and time based firewall blocking based on IP flood attacks– Detection of known IP sniffers and DDOS attack generators


Advanced Call Routing (‘Softswitch’)• Advanced GUI or XML Dialplan

– Route calls based on any sip header or DID or IP– Nested dialplan support with advanced regex matching

• Database Routing– Routing based on remote database lookup using HTTP/HTTPS– Routing based on ODBC database connection– Mongo DB support

• Load Balancing– Weighed or round robin load balancing between multiple SIP interfaces within a domain

• Least Cost Routing– Support for local LCR database. GUI LCR Import/export.

• DNS/SRV Routing• DHCP Options


© 2014 Sangoma Technologies

Troubleshooting• GUI Error Reporting and Notification

– GUI dashboard with time based graphing– System, session, capacity errors– Error message counts

• PCAP Tracing– Ability to trace both signaling and media on the SBC. Self contained No need to use external port mirrors or hub– Decode PCAP files using Wireshark– Huge disk space to store large circular PCAP buffer for long term debugging

• RTCP Search– Search for calls with bad RTCP thresholds. Email notifications on each bad RTCP call.

• SSH and CLI Console– Ability to perform real time log analysis and tracing on the console– Multiple screen support

• Logging– Extensive logging per call tagged using UUID– Remote syslog support

• Hardware Crash Protection– Automatic reboot on system lockup or HW fault

22

Business Applications and Use Cases

Vega Series SBC

Enterprise SIP Trunking


Vega eSBC

ITSP SIP SIP

IP-PBX

IPSIP

DMZ Deployment

Vega eSBC

ITSP SIP SIP

IP-PBX

IPSIP

Direct Deployment on Public IP address

Secure Access Control for Remote Users or Telecommuters


Vega eSBC

ITSP SIP SIP

IP-PBX

IPSIP

ExternalFW/NAT

InternalFW

Home Office,Mobile Users, Telecommuters

SIP

Vega eSBC:• Pass-through SIP registration on IP-PBX• Remote FW/NAT traversal• Call Admission Control• Topology Hiding• TLS and SRTP encryption• No VPN required

Ext 101

Ext 102

Multi-Site Consolidation


SBC:• Remove Multi-Sites PRIs• Performs SIP Security Functions• SIP Harmonization• Media Harmonization• Intelligent Call Routing

• Sophisticated Dial Plans

Vega eSBC

ITSP SIP SIP

IP-PBX

IPSIP

IP-PBX

IP-PBX

SIP

SIPWAN

WAN

Carrier SBC For Hosted PBX• Advantages

– Known demarcation point– Reduces interoperability issues/resource with core– Transcoding if required


IP Phones

IP Network

VoIP ServiceProvider

LAN VoIP

Multi-TenantIP PBX

Legacy PBX Migration to Microsoft Lync


SBC:• Performs SIP Security Functions• UDP/TCP Translation• SIP Harmonization

• Media Harmonization

• Intelligent Call Routing• Active Directory Routing• Unified Dial Plan

Vega eSBC

MediationServer

LyncServer

LyncUser

ITSP SIP

SIP

SIP

IP-PBX

ActiveDirectory

Microsoft Lync Transition with Analog Lines


SBC:• Performs SIP Security Functions• UDP/TCP Translation• SIP Harmonization

• Media Harmonization

• Intelligent Call Routing• Active Directory Routing• Unified Dial Plan

Vega eSBC

MediationServer

LyncServer

LyncUser

ITSP SIP

SIP

SIP

Vega 5000

ActiveDirectory

Analog5000

SIP Signaling Conversion

• Convert SIP over TCP to SIP over UDP• Some devices require SIP/TCP

– e.g. Microsoft Lync


Carrier/Service Provider Applications and Use

Cases

NetBorder Series SBC

Carrier SBC for SIP Carrier

ITSP

Softswitch

SBCBroadban

d

NAT/FW

NAT/FW

SIPSIP

SIP SIP

ATA

NAT/FW

SIP

SIP

SBC:• Performs SIP Security Functions• Peering with Other SIP Providers• SIP Harmonization• Media Harmonization• Far End NAT Traversal• Call Admission Control

Residential

Residential

SOHO

SIP


SIP Trunking


This NetBorder SBC

protects the ITSP’s network

SIP Network Peering/IP Carrier Interconnect

• Use IP for inter-carrier links• No TDM conversion required:

– Decrease complexity– Better voice quality, less delay, less transcoding


Carrier Interconnect Mediation• Secure carrier network• Normalize SIP messaging (easy interop)• Transcoding between carriers


SBC Load Balancing and Failover Techniques

Typical Service Provider SBC Deployment

• Hosted PBX Service• SBC Protects SP’s Network; performs far end NAT traversal, etc.• Each VoIP phone sends all SIP protocol messages to SP’s

Softswitch via SBC (phone’s outbound proxy settings) • SBC is critical; if it fails no service for 1000s of users


Internet

Softswitch

RouterNATFW

ServiceProvider

Load Balancing SBCs with DNS SRV


Internet

Softswitch

RouterNATFW

ServiceProvider

DNS Server

sbc110.10.0.10

sbc210.10.0.20

DNS SRV Record Query for ‘carrier.com’

_sip._udp.carrier.com 60 IN SRV 10 50 5060 sbc1.carrier.com_sip._udp.carrier.com 60 IN SRV 10 50 5060 sbc2.carrier.com

DNS A Record Query

sbc1.carrier.com = 10.10.0.10sbc2.carrier.com = 10.10.0.20

Same priority and weight entries:• sbc1 and sbc2 would each get

50% of the traffic load• If one SBC becomes

unavailable, remaining machine takes the load

1

2

Domain: carrier.com

Failover SBCs with DNS SRV


Internet

Softswitch

RouterNATFW

ServiceProvider

DNS Server

sbc110.10.0.10

sbc210.10.0.20



DNS A Record Query


• Different Priorities• Lower Priority tried first:

sbc1.carrier.com•If sbc1.carrier.com unavailable:

sbc2.carrier.com

1

2

Domain: carrier.com

DNS SRV: Countless Other Scenarios• DNS SRV records not limited to 2 lines• Could implement several scenarios:

– M-ways load balancing– M-ways load balancing; N-way failover

• Example:

• _sip._udp.carrier.com 60 IN SRV 10 60 5060 sbc1.carrier.com • _sip._udp.carrier.com 60 IN SRV 10 20 5060 sbc2.carrier.com • _sip._udp.carrier.com 60 IN SRV 10 10 5060 sbc3.carrier.com • _sip._udp.carrier.com 60 IN SRV 10 10 5060 sbc4.carrier.com • _sip._udp.carrier.com 60 IN SRV 20 0 5060 sbc5.carrier.com

• The first 4 SBC would share the load at 60%, 20%, 10% and 10% respectively• If the first 4 SBCs should become unavailable, sbc5 would take the load


SBC Walkthrough

SBC Walkthrough

• For the walkthrough, we will cover the following topics:– Logging into your SBC– System Overview and Services– Configuring your Signaling Interfaces– Configuring your Media Interfaces


The SBC System Status


Signaling Interfaces – Highlights


• Each interface used for signaling is displayed, with the ability to edit that interface• User can create multiple virtual interfaces

• Multiple virtual interfaces can be created and applied to a single physical interface• User can create VLAN interfaces if the SBC is sitting in a specific VLAN

• Multiple VLAN interfaces can be applied to a single physical interface

Signaling Interfaces – Configuring an Interface


• Simplistic Configuration of signaling interface• Select from either a static IP assignment or a dynamic DHCP assignment• Apply an appropriate IP address and Network Mask to the interface• Options can be an variation of any Linux Ethernet ethtool options

• Ex. speed 1000 duplex full autoneg off

Signaling Interfaces – sngdsp Interface

• The SNGDSP interface is a special interface within the SBC

• The interface controls all the interactions between the media adapters and the SBC

• When configuring the SBC, the sngdsp interface must be on a “non-routable” network or a WAN/DMZ IP address– IP address configuration depends on whether you will be

configuring the media adapters in exposed or hidden mode. This will be explained when looking at the media interfaces.


Media Interfaces – Highlights


• Outlines the way the media interfaces are configured and details information on each Media DSP adapter

• Media Server configuration is the method in which the DSPs are configured • Hidden mode hides the DSPs from the environment• Exposed mode exposes the DSPs to the environment. If in exposed mode, each DSP must have

a routable IP address configured.• Software mode identifies that no SngDsp interface is installed. Transcoding and TLS/SRTP are

disabled. This is found only in the software only version of the SBC.• Each SngDsp interface will come with preinstalled adapters. This will depend on which hardware

version of the SBC is purchased.• The DSP version, MAC address, and assigned IP and RTP ports will be displayed. You can edit

each individually if required.

Media Interfaces – Configuration


First Practical


• Log in to SBC default details:– User : root– Password: sangoma– Default IP address 192.168.2.2

• Configure Signaling & Media Interfaces– Structure of WebUI

Expected Break for Lunch


Recap of Session 1

What is an SBC and why have one?• SBCs are installed at the edge of VoIP Networks to facilitate

end to end VoIP transmission without compromising network security

• Essential for several reasons:– New security issues introduced with SIP protocol– Fix interoperability issues

• SBC are typically implemented as Back to Back User Agents (B2BUA)– All SIP and media (voice) traffic transit through SBCs


Vega Enterprise SBC• Appliance

– 25-250 Sessions– H/W DSP acceleration– 1U/2 x 1 GE ports

• Software Version– 25-500 Sessions/Self-Contained ISO– VM requirements


• Software/Hybrid Version –UNIQUE– 25-500 Sessions/Self-Contained ISO– VM requirements


– H/W DSP acceleration


D150

NetBorder Carrier SBC

• Appliance– 250-4000 Sessions– H/W DSP acceleration– 1U/2 x 1 GE ports– RAID 1


Failover SBCs with DNS SRV


Internet

Softswitch

RouterNATFW

ServiceProvider

DNS Server

sbc110.10.0.10

sbc210.10.0.20



DNS A Record Query


• Different Priorities• Lower Priority tried first:

sbc1.carrier.com•If sbc1.carrier.com unavailable:

sbc2.carrier.com

1

2

Domain: carrier.com

Signaling Interfaces – Highlights


• Each interface used for signaling is displayed, with the ability to edit that interface

• User can create multiple virtual interfaces• Multiple virtual interfaces can be created and applied to a single

physical interface• User can create VLAN interfaces if the SBC is sitting in a specific VLAN

• Multiple VLAN interfaces can be applied to a single physical interface

Signaling Interfaces – sngdsp Interface

• The SNGDSP interface is a special interface within the SBC

• The interface controls all the interactions between the media adapters and the SBC

• When configuring the SBC, the sngdsp interface must be on a “non-routable” network or a WAN/DMZ IP address– IP address configuration depends on whether you will be

configuring the media adapters in exposed or hidden mode. This will be explained when looking at the media interfaces.


Media Interfaces – Highlights


• Outlines the way the media interfaces are configured and details information on each Media DSP adapter

• Media Server configuration is the method in which the DSPs are configured • Hidden mode hides the DSPs from the environment• Exposed mode exposes the DSPs to the environment. If in exposed mode, each DSP

must have a routable IP address configured.• Software mode identifies that no SngDsp interface is installed. Transcoding and

TLS/SRTP are disabled. This is found only in the software only version of the SBC.• Each SngDsp interface will come with preinstalled adapters. This will depend on which

hardware version of the SBC is purchased.• The DSP version, MAC address, and assigned IP and RTP ports will be displayed. You

can edit each individually if required.

SBC Call Processing Components

Conceptual Overview

Conceptual Overview


Carrier SIP Trunk

External SIP Profile

External Media Profile

Domain Profile

Inbound Call Routing

Internal SIP Trunk

Internal SIP Profile

Internal Media Profile

Outbound Call Routing

SIP ProfilesIntroduction to

Carrier SIP Trunk

SIP Profile Overview


1External

SIP ProfileExternal

Media Profile

Domain Profile


Internal SIP Trunk




What is a SIP Profile?• A SIP Profile defines a SIP interface in the SBC • The SIP profile defines a set of SIP attributes that are

associated to the SIP interface on the SBC• The SIP Profile is used as a portal external endpoints or

Trunks which connect to the SBC• An IP interface address and port are bound exclusively to the

SIP Profile• As part of the config call routing plans, domain profiles,

media profiles, and SIP trunks are bound to the SIP profiles


SIP Profile Overview• Any call, whether inbound or outbound, gets

processed within a SIP profile• Profile has a choice where to send the call based on

call routing• In general, there is always a minimum of 2 profiles

within an SBC– One is internal– One is external


SIP Profiles


• SIP Profiles can be managed from the SIP Profile menu option which is contained under Configuration

• A list of all the configured SIP Profiles is listed, with the ability of modifying a profile or removing it entirely

• The ability to sort and search is also available to users who have multiple SIP Profiles• This would happen if the SBC is being used in a multi-tenant or carrier situation

Domain ProfilesIntroduction to

Carrier SIP Trunk

Domain Profile Overview


12



Domain Profile


Internal SIP Trunk




What is a Domain Profile?• A domain, or a SIP realm, is a component within SIP which is

used to authenticate users within the SIP Registration process• Domain profiles are used to define the way users will

authenticate with the SBC– Local authentication is used when users will register with the SBC– Upper registration is used when users will register to a softswitch or a

IP-PBX through a SBC• This enables topology hiding so that no one outside of the corporate network

knows about the equipment sitting behind the SBC

• If using IP authentication, you will not require a domain profile


Domain Profile Overview• When a registration arrives into a SIP profile, the SIP profile

then sends off that request to the domain profile which is bound to it

• The domain profile details where to send the registration request– Local authentication challenges the registration from within the SBC

• User information is stored within the SBC

– Upper registration proxies the request to an IP-PBX or softswitch so that the registration request can be validated

• User information is stored within the IP-PBX or the softswitch


Domain Profiles

• Domain profiles can be managed from the Domain Profile menu option within the Configuration menu

• Domain profiles can be either a domain name (ex. Sangoma.com) or an IP address (ex. 10.82.1.254)

• The name of the domain profile must correspond to the way users register to either the SBC

– If the users will register to a domain name, the name of the domain profile MUST be that domain name (i.e. [email protected])

– If the users will register with an IP address, the name of the domain profile MUST be the IP address (i.e. [email protected])


mailto:[email protected]

mailto:[email protected]

Media ProfilesIntroduction to

Carrier SIP Trunk

Media Profile Overview


12External

SIP ProfileExternal

Media Profile

Domain Profile


Internal SIP Trunk




What is a Media Profile?

• A media profile is a list of attributes which defines what audio codecs are used on a per call basis

• It also describes how DTMF (Dual Tone Multiple Frequency) will be handled within the SIP profile


Media Profile Overview• The media profile deals with the codec negotiation which occurs

before a call is established• Media profiles have a list of codecs within them, and detail how

the process will occur– Either the remote end will choose the codec or the SBC will choose the

codec

• Codec negotiation is usually performed within the SIP SDP (Session Description Protocol)– The SDP is usually found in the SIP 200 OK message which is sent prior

to call establishment and the flow of RTP


Media Profiles

• Media profiles are used to determine which audio codecs will be used with SIP profiles– Multiple codecs are only available on the Carrier, Enterprise, or Hybrid SBCs– The software version of the SBC will only allow G.711

• You can also choose the codec negotiation mode, DTMF (Dual-Tone Multi Frequency) mode, and whether to enable/disable silence suppression

• By default, only 1 profile is created– Multiple profiles can be created– No limit to the amount of profiles– You can create an inbound profile and outbound profile, then use that in your SIP profiles– Possibilities are endless…


Media Profile – Codecs• An audio codec is a program implemented as an algorithm that

compresses and decompresses digital audio data• 5 codecs can be configured per media profile

– 10 different codecs to choose from with multiple variations of each codec– Codecs available:

• G.711 PCMU • G.711 PCMA• G.729• AMR• iLBC• GSM• G.722• G.722.1• G.723• G.726


SIP TrunksIntroduction to

Carrier SIP Trunk

Internal SIP Trunk


SIP Trunk Overview1

4


Domain Profile





3

2


What is a SIP Trunk?

• SIP trunks are used to create a communication path between 2 SIP aware endpoints

• Trunks can be used to communicate with SIP carriers or with IP-PBXs– It is the description of how the SBC will

communicate with that endpoint– Example: IP address, port, etc.


SIP Trunk Overview

• In a scenario where a call is trying to be established, the SIP profiles need to know about the endpoints

• SIP trunking allows for the profiles to be aware of where calls may originate from– Calls will arrive from a trunk, then are processed

within the profile, then are sent to call routing in order to redirect to a different trunk


SIP Trunks

• SIP trunks attach to SIP profiles from within the trunk configuration

• Multiple SIP trunks can be created– A SIP profile can control multiple SIP trunks, but a SIP

trunk can only be bound to a single SIP profile


Call RoutingIntroduction to

What is Call Routing• Call routing is the process used to route telephone

calls across a telephony network• The process is the same whether calls are made

between two phones in the same locality, or across two different continents


Call Routing Concepts


Carrier SIP Trunk



Domain Profile

Internal SIP Trunk


Ingress Header Manipulation

Ingress Header Manipulation



Egress Header Manipulation


Egress Header Manipulation

1

2

4

3

Call Routing Concepts• There are three concepts to call routing:

1. Condition• The outcome this routing rule is addressing • The condition statement is used to determine how the call will be dealt if the rule

turns out to be true or false• Example: Check the destination number is +19054741990

2. Action to be performed if true• What action will be performed if the condition is found to be true?• Example: Bridge to a different SIP trunk

3. Action to be performed if false• What action will be performed if the condition is found to be false?• Example: Send the originator a 503 service unavailable message


Call Routing Explained• Routing rules are created in order to direct calls received from one

interface, and bridge it out to the next interface– SIP profiles or SIP trunks are used to bridge calls

• Routing rules can be as simple as bridging between trunks, or as complicated as choosing from a different carrier due to costs of routing

• There are two different methods for creating routing rules within the SBC– Basic Routing– Advanced Routing


Basic Call Routing• Basic call routing uses the graphic user interface of the SBC

to allow users to create routing rules• It is modeled so that anyone would be able to create almost

any type of scenario without the need to learn XML• Each basic dialplan can have multiple rules associated with it

– Each rule deals with a specific condition which needs to be met– You can program the rule to continue to the next rule if it passes or

fails


Basic Dialplan Aggregated View

• First section deals with default parameters for that particular dialplan• Second section deals with the specific rules which will be processed

within the dialplan– Each rule is described based on the selections chosen within the rule

configuration


Advanced Call Routing• For advanced users, there is a way to build dialplans using the advanced

call routing engine• Advanced call routing is based on XML• There is no need to build multiple rules

– All rules are added into a single XML file– Rules are separated by the different conditions

• There are different editors built into the advanced dialplan that a user may choose from:– Standard text editor– Vim editor– Emacs editor


Advance Call Routing


Walkthrough

SBC Walkthrough

• Will cover the following configurable sections within the SBC:– SIP Profiles– SIP Trunks– Domain Profiles– Call Routing


Questions?

THANK YOU!

Technology

Sangoma SBC Training Presentation