Intertex - Telia
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Transcript Intertex - Telia
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
[email protected]
© 2010 Intertex Data AB
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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 40-50 K (up to) 8 lines SMB PBXs (only in Sweden)
© 2011 Intertex Data and Ingate Systems
Confidential
2
A Good Triple Play Network was Available
Architecture deployed by carriers
to assure QoS for and control of
Voice, TV and other multimedia.
Internet
VoIP
Mng
IP-TV
Let’s make
VoD
PVC2
PVC1
PVC3
SIPv1
PVC4
VLANs or ADSL
Virtual Circuits
for SIP Trunking
of IP-PBXs
ADSL Modem
“Triple Play”
LAN
ATA
© 2011 Intertex Data and Ingate Systems
Confidential
3
So Just Hook up the IP-PBX to the VoIP Pipe…
Telephony
TV
SIP Trunk
Interface
Internet
PBX with
system
phones
REQUIREMENTS:
As good as before (as TDM)
8 simultaneous calls, 10 – 100
numbers
© 2011 Intertex Data and Ingate Systems
Confidential
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Telia SIP Connection, Business Broadband
IP-PBX
OVERVIEW
ADSL-modem
Telia SIP Connection
Triple play - Bridged
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…
1) Be DHCP client
2) Register all accounts (all DID numbers) to Telia’s SIP platform
Cut and translated from Telia presentation
5
There were some ISSUES
IP-PBX
ADSL-modem
Nnn SIP Connection
Triple play - Bridged
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
6
…and a few more ISSUES
IP-PBX
ADSL-modem
Nnn SIP Connection
Triple play - Bridged
Registration
Signaling
Telephony gateways
Load balancing
Remote administration
Port 1
Router
NAT
Internet
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
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Using the IX78 E-SBC solved those issues, but…
8
There are more things to consider…
PSTN
An E-SBC should provide:
1) NAT/Firewall Traversal – Must NAT to same address space!
2) Basic SIP and Network Interoperability - E.g.
SIP Trunking
Provider Network
Authentication, Registrations, UDP/TLS/TCP, Dynamic IP address, etc.
SIP System
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
SIP Trunk
1) 2) 3) 4) 5)
IX78
IPPBX
2) 3) 4) 5)
2) 3) 4) 5)
SIP Trunk Interface
Modern IP-PBXs are of
this type. Media goes
directly between phone
and SIP Trunk.
PBX with
system
phones
IPPBX
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
VoIP & Data LAN
Data LAN only
PBX Type 1
Signaling:
Media:
PBX Type 1.5
PBX Type 2
And then make it easy to install and configure
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Confirmed Interoperability: Ingate & Intertex
SIP Trunk Providers
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
IP-PBXs
Sylantro
SER
NSN
More in pipeline…
© 2011 Intertex Data and Ingate Systems
Confidential
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....
The SIP Trunking Installation Wizard
jkjjk
© 2011 Intertex Data and Ingate Systems
Confidential
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!
© 2011 Intertex Data and Ingate Systems
Confidential
13
Into the TeliaSonera Lab!
Testing, integrating with
management system
(existing TR-069 ACS),
creating a service…
…and checking new
PBXs
PBXs
PBXs
© 2011 Intertex Data and Ingate Systems
Confidential
The IX78 Supports Many WAN Layer 2 and Layer 3 Architectures
with QoS Separated WAN Interfaces (inherited from it’s triple play capabilities)
E.g. Telia
E.g. Telia
Internet
IP-TV
VoD
Internet
IMS
IP-TV
VoIP
VoD
IMS
VoIP
PVC1
VLAN1
PVC3
PVC2
ADSL
Virtual LANs (VLAN)
Ethernet
Private Virtual Circuits
E.g. B2
VLAN3
VLAN2
E.g. BT
Internet
IP-TV
VoD
IMS
IP-TV
Priority2
VoIP
VoD
Internet
Priority3
IMS
VoIP
Priority1
WAN1
WAN2
Ethernet
WAN3
IP QoS Separated Subnets
ADSL or Ethernet
IP Level QoS
The Intertex IX78 Supports All of these Architectures!
© 2011 Intertex Data and Ingate Systems
Confidential
15
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:
Signaling
6%
Signaling
3%
Free CPU
32%
Media
30%
Free CPU
67%
24 calls, 5 min/call, 20 ms packets
Media
62%
50 calls, 5 min/call, 20 ms packets
60 simultaneous calls without MOS degeneration were reached!
© 2011 Intertex Data and Ingate Systems
Confidential
16
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
IPPBX
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)
© 2011 Intertex Data and Ingate Systems
Confidential
IPPBX
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Trunk-side Parameters (B2BUA Mode)
© 2011 Intertex Data and Ingate Systems
Confidential
18
PBX-side Parameters (B2BUA Mode)
© 2011 Intertex Data and Ingate Systems
Confidential
19
Registration, Call Routing, CallerID (B2BUA Mode)
© 2011 Intertex Data and Ingate Systems
Confidential
20
Support for UC LAN and Multimedia Terminals as well as Remote Users
Public
Internet
SIP Trunking
Provider
PSTN
SIP System
Remote
Users
Intertex IX78
Demarcation point of
service and bringing SIP
communication to the LAN
IP-PBX
Data & VoIP LAN
Soft Clients and Multimedia Terminals
© 2011 Intertex Data and Ingate Systems
Confidential
21
Usage Together With Existing Firewall Also Important
PSTN
Public
Internet
PSTN
SIP Trunk
Provider
Public
Internet
SIP System
SIP Trunk
Provider
SIP System
Bridge for Existing
NAT/ Firewall
(non SIP aware)
IPPBX
SIParator®
WAN
SIParator®
IPPBX
Data & VoIP LAN
Data & VoIP LAN
If common IP pipe, the existing
firewall must restrict bandwidth
usage to allow sufficient voice
bandwidth. Often problematic.
WAN SIParator mode allows the
Ingate or Intertex to control data
usage on the Pipe to assure
sufficient voice bandwidth!
© 2011 Intertex Data and Ingate Systems
Confidential
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.
© 2011 Intertex Data and Ingate Systems
Confidential
SIP Capable Firewalls
See us at ITEXPO Room A208!
Intertex Data AB
Ingate Systems Inc.
www.intertex.se
[email protected]
Rissneleden 45
SE-174 44 Sundbyberg
Sweden
sip:[email protected]
Tel: +46 8 6282828
www.ingate.com
[email protected]
7 Farley Road Hollis
NH 03049
United States
Ph: +1 (603) 883-6569
Ph Sweden: +46 8 6007750
© 2011 Intertex Data and Ingate Systems
Confidential
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