Lessons Learned Across the Pond SIP Trunking Towards the All-IP Phone Network © 2010 Intertex Data...

Lessons Learned Across the PondSIP Trunking Towards the All-IP Phone Network

© 2010 Intertex Data AB 1

Prepared for: INTERNET TELEPHONY ConferenceIngate’s SIP Trunking SummitMiami, February 2011

By: Karl Erik Ståhl President & CEO Intertex Data ABChairman Ingate Systems [email protected]

© 2011 Intertex Data and Ingate Systems Confidential 2

Towards the All-IP Telephone Networkat Sweden’s Telco, TeliaSonera

The TDM network will be too expensive to maintain. ISDN (BRI) subscriber lines to be scratched first.

But there are the 40-50 K (up to) 8 lines SMB PBXs (only in Sweden)


A Good Triple Play Network was Available

MngMng

InternetInternet

PVC1

IP-TV

VoD

IP-TV

VoD

VoIP VoIP

PVC2PVC3

PVC4

LAN

Architecture deployed by carriers to assure QoS for and control of Voice, TV and other multimedia.

ADSL Modem“Triple Play”

VLANs or ADSL Virtual Circuits

Let’s make

SIPv1 for SIP Trunking

of IP-PBXs

ATA


Internet

Telephony

So Just Hook up the IP-PBX to the VoIP Pipe…

TV

PBX with PBX with system system phonesphones

SIP

Tru

nk

Inte

rfa

ce

REQUIREMENTS:

As good as before (as TDM)

8 simultaneous calls, 10 – 100 numbers

5

Telia SIP Connection, Business Broadband OVERVIEW

Internet

Telia SIP ConnectionRegistrationSignalingTelephony gatewaysLoad balancing

ADSL-modemTriple play - Bridged

Port 3

Port 1

VoIP

IP-PBX

Internet and VoIP travel over separate channels (PVC) and are delivered on separate physical portsDifferent subnets for Internet and VoIPDynamic public IP-addresses assigned to ports 1 and 3Prioritized capacity for eight concurrent calls on the VoIP channelOne DID telephone number corresponds to one SIP account

SO THE PBX WITH A SIP TRUNKING INTERFACE JUST HAS TO…

1)Be DHCP client

2)Register all accounts (all DID numbers) to Telia’s SIP platform

Cut and translated from Telia presentation

6

There were some ISSUES

Internet

Port 3

Port 1

VoIP

• If the IP-PBX can’t act as a DHCP client, some type of NAT-router must be used between the IP-PBX and the ADSL modem.• And the IP-PBX must be able to register all SIP accounts on the Nnn platform.• Some method has to be used in combination with the router for SIP traversal:

• STUN - Simple Traversal of UDP through NATs (Network Address Translation)• SIP-ALG – Application Layer Gateway

• But, remote administration over the same ADSL access is not possible…

Add RouterNAT?

LAN WAN

Remote administration


IP-PBX Nnn SIP ConnectionRegistrationSignalingTelephony gatewaysLoad balancing

Cut and translated from Telia presentation

Cut and translated from Telia presentation 7

…and a few more ISSUES

Internet

Port 3

Port 1

VoIP

More issues that have caused some headache:• IP-PBX’s that only accepts calls from known servers (incompatible with load balancing)• IP-PBX’s that only can register one account but expects incoming calls to all DID

telephone numbers, using this single account• Routers which can’t handle fragmented IP packets• Remote administration over the same ADSL access

RouterNAT

LAN WAN

Remote administration


IP-PBX Nnn SIP ConnectionRegistrationSignalingTelephony gatewaysLoad balancing

Out of 10 selected PBXs, none could be used straight of!

8

Using the IX78 E-SBC solved those issues, but…

There are more things to consider…

Data LAN only

PBX with PBX with system system phonesphones

PBX Type 1.5

VoIP & Data LAN

PBX Type 2

IPIP-- PBXPBX

Few PBXs are of this type. Asterisk with firewall (IPtables /NETfilter) can be compiled and configured this way, but requires a lot.

An E-SBC should provide:1) NAT/Firewall Traversal – Must NAT to same address space!

2) Basic SIP and Network Interoperability - E.g. Authentication, Registrations, UDP/TLS/TCP, Dynamic IP address, etc.

3) SIP Repair - E.g. Call Transfer, Fragmented packets, Bugs, etc. 4) Features - E.g. Remote Users, Administration (remote and local)

5) Security - LAN/PBX/VoIP network protection, Service attack protection

VoIP & Data LAN

IPIP-- PBXPBX

PBX Type 1

Modern IP-PBXs are of this type. Media goes directly between phone and SIP Trunk.

SIP Trunk Interface

Signaling:Media:

SIP Trunk

PSTNSIP Trunking

Provider NetworkGW

SIP System

2) 3) 4) 5)2) 3) 4) 5)IX78

1)1) 2) 3) 4) 5)2) 3) 4) 5) 2) 3) 4) 5)2) 3) 4) 5)

10

And then make it easy to install and configure

© 2011 Intertex Data and Ingate Systems Confidential

Confirmed Interoperability: Ingate & IntertexSIP Trunk Providers IP-PBXs

SIP Trunk

Compliant with

Aastra Aastra/Ericsson MX One Adtran UC Server Digium/Asterisk Avaya Aura Avaya IP Office Avaya SES/CM Avaya QE Brekeke Broadsoft Cisco Fonality HP/3Com -VCX Innovaphone Interactive Intelligence Iwatsu LG Nortel Microsoft OCS Mitel NEC / Sphere Nortel BCM Nortel SCS Objectworld Panasonic Samsung SER Shoretel Siemens SIP-Gear SwyxMore in pipeline....

360 Networks Airespring AT&T BandTel Bandwidth.com Broadvox BT (British Telecom) Cablevision Cbeyond Cellip Comm Partners Cordia Corporation Deltacom Excel Switching Gamma Telecom GEOS Global Crossing IP-Only Nectar Level 3 Netlogic Netsolutions

Nexvortex Nuvox O1 One Communications Paetec Primus RNK Telecom Skype TDC Telavox Tele2 Tele Pacific Teletek TeliaSonera Toplink Tritel VoEX Voice Flex VoIP Unlimited Voxbone Voxitas XeloQMore in pipeline...

Carrier Equipment Acme Packet Broadsoft Genband Sonus

Sylantro SER NSN More in pipeline…


The SIP Trunking Installation Wizard

jkjjk


All worked fine – But, time to make SIPv2! New SIP IMS platform

Will take over generally for the future Higher scale But “more complex” SIP interface

More IP delivery networks ADSL2+ AnnexM: Triple play as before FiberLAN: 100 Mbps Ethernet triple play (VLAN tagged) Prolane: Internet with priority VoIP channel Internet: Telia’s SIP Trunking over other providers Internet access

Up to 60 simultaneous calls per trunk group (8 in SIPv1)

CPE / E-SBC comes with the service, owned by Telia

Provisioning and management by Telia Reused ACS (TR-069 management system) for residential Combined with Intertex PBX selection Wizard

What is required from the CPE / E-SBC? Intertex IX78 still the choice!


Into the TeliaSonera Lab!

Testing, integrating with management system (existing TR-069 ACS), creating a service…

…and checking new PBXs

PBXs

PBXs


The IX78 Supports Many WAN Layer 2 and Layer 3 Architectures with QoS Separated WAN Interfaces (inherited from it’s triple play capabilities)

The Intertex IX78 Supports All of these Architectures!

Private Virtual Circuits

E.g. Telia

InternetInternet

ADSL

PVC1

IP-TV

VoD

IP-TV

VoD

IMS

VoIP

IMS

VoIP

PVC2 PVC3

E.g. Telia

InternetInternet

Ethernet

VLAN1

IP-TV

VoD

IP-TV

VoD

IMS

VoIP

IMS

VoIP

VLAN2 VLAN3

Virtual LANs (VLAN)

E.g. B2

InternetInternet

Ethernet

WAN1

IP-TV

VoD

IP-TV

VoD

IMS

VoIP

IMS

VoIP

WAN2 WAN3

IP QoS Separated Subnets IP Level QoS

E.g. BT

InternetInternet

ADSL or Ethernet

Priority3Priority2 Priority1

IMSVoIP

IP-TVVoD


Performance and Call Handling Capacity

Over 50 simultaneous calls (20 ms voice packets) carrying media

Call rate of 8 calls/s in proxy mode and 3 calls/s in B2BUA mode. (more than required to support 50 simultaneous calls)

Up to 255 registrations. SIP end-points can be more.

CPU Usage:

50 calls, 5 min/call, 20 ms packets

Free CPU32%

Media62%

Signaling6%

24 calls, 5 min/call, 20 ms packets

Free CPU67%

Media30%

Signaling3%

60 simultaneous calls without MOS degeneration were reached!


IMS and More Required use of B2BUA Mode

Proxy Mode IP-PBX talks to Service

Registration/Authentication model must match

Little configuration in the IX78

Service credentials in the PBX

B2BUA Mode (Proxy still doing the basics) IP-PBX only talks to the IX78

Wider separation between PBX and Service

Service Credentials only in the IX78

More SIP Normalization possibilities (e.g. REFER)

Any new operator service platform only requires IX78 reconfiguration (the PBX configuration can remain)

IP- PBX

IP- PBX


Trunk-side Parameters (B2BUA Mode)


PBX-side Parameters (B2BUA Mode)


Registration, Call Routing, CallerID (B2BUA Mode)


PSTNPublic

Internet

SIP Trunking Provider

GWSIP System

Data & VoIP LAN

IP-PBX

Demarcation point of service and bringing SIP communication to the LAN

Soft Clients and Multimedia Terminals

Intertex IX78

Remote Users

Support for UC LAN and Multimedia Terminals as well as Remote Users


Usage Together With Existing Firewall Also Important

PSTN

Public Internet

SIP Trunk Provider GW

SIP System

IP- PBX

NAT/ Firewall

Data & VoIP LAN

If common IP pipe, the existing firewall must restrict bandwidth usage to allow sufficient voice bandwidth. Often problematic.

PSTN

Public Internet

SIP Trunk Provider GW

SIP System

IP- PBX

NAT/ Firewall

Bridge for Existing NAT/ Firewall (non SIP aware)

Data & VoIP LAN

WAN SIParator mode allows the Ingate or Intertex to control data usage on the Pipe to assure sufficient voice bandwidth!

WAN SIParator®SIParator®

22


Jonas Östergren, TeliaSonera, Interviewed Live from Sweden

Using Omnitor application Allan eC:

Voice: G.722 wide band codec

Video: H.264 300kbps

Real-time text: RFC4103

Using standard SIP over the Internet.


SIP Capable Firewalls

Ingate Systems [email protected] Farley Road HollisNH 03049United StatesPh: +1 (603) 883-6569Ph Sweden: +46 8 6007750

Intertex Data [email protected] 45 SE-174 44 SundbybergSwedensip:[email protected]: +46 8 6282828

See us at ITEXPO Room A208!

Lessons Learned Across the Pond SIP Trunking Towards the All-IP Phone Network © 2010 Intertex Data...

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Transcript of Lessons Learned Across the Pond SIP Trunking Towards the All-IP Phone Network © 2010 Intertex Data...