1. Lessons Learned Across the Pond
SIP Trunking Towards the All-IP Phone Network
Prepared for: INTERNET TELEPHONY Conference
Ingate’s SIP Trunking Summit
Miami, February 2011
By: Karl Erik Ståhl
President & CEO Intertex Data AB
Chairman Ingate Systems AB
© 2010 Intertex Data AB 1

2. Towards the All-IP Telephone Network
at 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 K (up to) 8 lines SMB PBXs (only in Sweden) 2

3. A Good Triple Play Network was Available
Internet
Architecture deployed by carriers to assure QoS for and control of Voice, TV and other multimedia.
VoIP
Mng
IP-TV
VoD
Let’s make
SIPv1
for SIP Trunking
of IP-PBXs
PVC3
PVC2
PVC1
PVC4
VLANs or ADSL Virtual Circuits
ADSL Modem
“Triple Play”
LAN
ATA

4. So Just Hook up the IP-PBX to the VoIP Pipe…
Telephony TV
Internet
PBX with
system
phones
SIP Trunk
Interface 
REQUIREMENTS:
As good as before (as TDM)
8 simultaneous calls, 10 – 100 numbers 4

5. Telia SIP Connection, Business Broadband OVERVIEW
Header: Relation
Telia SIP Connection, Business Broadband OVERVIEW
IP-PBX
ADSL-modem
Triple play - Bridged
Telia SIP Connection
Registration
Signaling
Telephony gateways
Load balancing
Port 1
Internet
VoIP
Port 3
Internet and VoIP travel over separate channels (PVC) and are delivered on separate physical ports
Different subnets for Internet and VoIP
Dynamic public IP-addresses assigned to ports 1 and 3
Prioritized capacity for eight concurrent calls on the VoIP channel
One DID telephone number corresponds to one SIP account
SO THE PBX WITH A SIP TRUNKING INTERFACE JUST HAS TO…
Be DHCP client
Register all accounts (all DID numbers) to Telia’s SIP platform
Cut and translated from Telia presentation

6. There were some ISSUES IP-PBX ADSL-modem Nnn SIP Connection Add Router
Header: Relation
There were some ISSUES
IP-PBX
ADSL-modem
Triple play - Bridged
Nnn SIP Connection
Registration
Signaling
Telephony gateways
Load balancing
Remote administration
Add
Router
NAT?
Port 1
Internet
VoIP
LAN
WAN
Port 3
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…
Cut and translated from Telia presentation

7. Header: Relation
…and a few more ISSUES
IP-PBX
ADSL-modem
Triple play - Bridged
Nnn SIP Connection
Registration
Signaling
Telephony gateways
Load balancing
Remote administration
Port 1
Internet
Router
NAT
VoIP
LAN
WAN
Port 3
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
Out of 10 selected PBXs, none could be used straight of!
Cut and translated from Telia presentation

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

9. There are more things to consider…
PSTN
SIP Trunking Provider Network GW
SIP System
An E-SBC should provide:
NAT/Firewall Traversal – Must NAT to same address space!
Basic SIP and Network Interoperability - E.g. Authentication, Registrations, UDP/TLS/TCP, Dynamic IP address, etc.
SIP Repair - E.g. Call Transfer, Fragmented packets, Bugs, etc.
Features - E.g. Remote Users, Administration (remote and local)
Security - LAN/PBX/VoIP network protection, Service attack protection
SIP Trunk
IX78 1)
2) 3) 4) 5)
2) 3) 4) 5)
2) 3) 4) 5)
VoIP & Data LAN
PBX Type 2
IP-
PBX
Few PBXs are of this type. Asterisk with firewall (IPtables /NETfilter) can be compiled and configured this way, but requires a lot.
VoIP & Data LAN
IP-
PBX
PBX Type 1
Modern IP-PBXs are of this type. Media goes directly between phone and SIP Trunk.
 SIP Trunk Interface  
Signaling:
Media:
Data LAN only
PBX with
system
phones
PBX Type 1.5

10. And then make it easy to install and configure10

11. Confirmed Interoperability: Ingate & Intertex SIP Trunk Providers IP-PBXs
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
Swyx
More 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
XeloQ
More in pipeline...
SIP Trunk
Compliant with
Carrier Equipment
Acme Packet
Broadsoft
Genband
Sonus
Sylantro
SER
NSN
More in pipeline…

12. The SIP Trunking Installation Wizard
jkjjk

13. 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! 13

14. Into the TeliaSonera Lab!
Testing, integrating with management system (existing TR-069 ACS), creating a service…
…and checking new PBXs
PBXs

15. The Intertex IX78 Supports All of these Architectures!
The IX78 Supports Many WAN Layer 2 and Layer 3 Architectures with QoS Separated WAN Interfaces (inherited from it’s triple play capabilities)
Private Virtual Circuits
E.g. Telia
Internet
ADSL
PVC1
IP-TV
VoD
IMS
VoIP
PVC2
PVC3
E.g. Telia
Internet
Ethernet
VLAN1
IP-TV
VoD
IMS
VoIP
VLAN2
VLAN3
Virtual LANs (VLAN)
E.g. B2
Internet
Ethernet
WAN1
IP-TV
VoD
IMS
VoIP
WAN2
WAN3
IP QoS Separated Subnets
IP Level QoS
E.g. BT
Internet
ADSL or Ethernet
Priority3
Priority2
Priority1
IMS
VoIP
IP-TV
VoD
The Intertex IX78 Supports All of these Architectures! 15 15

16. 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:
60 simultaneous calls without MOS degeneration were reached!

17. IMS and More Required use of B2BUA Mode
IP-
PBX
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

18. Trunk-side Parameters (B2BUA Mode)

19. PBX-side Parameters (B2BUA Mode)

20. Registration, Call Routing, CallerID (B2BUA Mode)

21. Support for UC LAN and Multimedia Terminals as well as Remote Users
SIP Trunking Provider
Public Internet GW
PSTN
SIP System
Remote Users
Intertex IX78
IP-PBX
Demarcation point of service and bringing SIP communication to the LAN
Data & VoIP LAN
Soft Clients and Multimedia Terminals

22. Usage Together With Existing Firewall Also Important
PSTN
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®
SIP Trunk Provider GW
Public Internet
SIP System
IP-
PBX
NAT/
Firewall
SIParator®
Data & VoIP LAN
If common IP pipe, the existing firewall must restrict bandwidth usage to allow sufficient voice bandwidth. Often problematic.

23. Jonas Östergren, TeliaSonera, Interviewed Live from Sweden
Using Omnitor application Allan eC:
Voice: G.722 wide band codec
Video: H kbps
Real-time text: RFC4103
Using standard SIP over the Internet.

24. SIP Capable Firewalls See us at ITEXPO Room A208!
Intertex Data AB
Rissneleden 45
SE Sundbyberg
Sweden
Tel:
Ingate Systems Inc.
7 Farley Road Hollis
NH 03049
United States
Ph: +1 (603)
Ph Sweden: