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)
© 2011 Intertex Data and Ingate Systems Confidential 3
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
© 2011 Intertex Data and Ingate Systems Confidential 4
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
ADSL-modemTriple play - Bridged
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
ADSL-modemTriple play - Bridged
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…
© 2011 Intertex Data and Ingate Systems Confidential
The SIP Trunking Installation Wizard
jkjjk
© 2011 Intertex Data and Ingate Systems Confidential 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!
© 2011 Intertex Data and Ingate Systems Confidential
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 15
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
© 2011 Intertex Data and Ingate Systems Confidential 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:
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!
© 2011 Intertex Data and Ingate Systems Confidential 17
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
© 2011 Intertex Data and Ingate Systems Confidential 18
Trunk-side Parameters (B2BUA Mode)
© 2011 Intertex Data and Ingate Systems Confidential 19
PBX-side Parameters (B2BUA Mode)
© 2011 Intertex Data and Ingate Systems Confidential 20
Registration, Call Routing, CallerID (B2BUA Mode)
© 2011 Intertex Data and Ingate Systems Confidential 21
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
© 2011 Intertex Data and Ingate Systems Confidential
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®
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© 2011 Intertex Data and Ingate Systems Confidential
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 24
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!
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