SIP

Clearvision Internet of SIP

SIP General Settings and IP PBX Compatibility

If a PBX is SIP compatible, then it will generally be immediately compatible with our SIP trunks. The most
important features for testing the compatibility of a PBX are these:
• Ability to set registration time at 120 seconds (halfs to 60s) where registration is required for inbound calls
• Ability to enter a username and password for authentication on outgoing calls
If your PBX is not SIP compatible (i.e a legacy PBX), then it may still be compatible with a SIP to PSTN Gateway
connected to the PBX, such as those made by Vega/Sangoma. Please call us for confirmation if you plan to
connect one to the service.

Protocols and encapsulation

SIP 2.0(RFC3261 and associated RFCs) for signalling
RTP for media encapsulation
UDP transport (default)
TCP transport (optional)
TLS | SRTP

SIP Trunk Gateways & Firewall
Configuration

st.sipconvergence.co.uk
Firewall configuration:
37.157.52.128/25 (37.157.52.130 – 37.157.52.254)
37.157.53.192/28 (37.157.53.194 – 37.157.53.206)
37.157.54.192/28 (37.157.54.194 – 37.157.54.206)
185.91.41.16/28 (185.91.41.18 – 185.91.41.32)

IP Ports

5060 UDP (SIP signalling)
5061 TCP TLS (SECURE SIP signalling)
10000-65335 UDP (RTP and SRTP stream)
TCP and UDP port 389 (LDAP) and 636 (LDAPS)

Outbound Call Authentication

Username/password authentication (default) OR IP authentication

DTMF (Tones)

RFC2833 (out-of-band)

Audio Codecs

G.711 a-Law, G.711 u-Law, G.722 (HD), GSM 6.10 FR (Full Rate) OPUS
Narrowband and OPUS Wideband

Video Codecs

H.264/MPEG-4, H.263, H.263+, VP8

Codec Transcoding

Audio – Supported
Video – Not Supported

Incoming number formats

E.164: +countrycode then number less leading zero, e.g. +441603904090 (SIP
trunk can be configured to strip the plus on the portal)

Outgoing number formats

UK format (e.g. 01603904090) or full E.164 format (e.g. +441603904090).

Voicemail to Email

The email servers send attachments as a *.wav
Email Server: relay01.tn19-f9.lon.gb.sipconvergence.co.uk (37.157.52.160)

If your PBX is not SIP compatible (i.e a legacy PBX), then it may still be compatible with a SIP to PSTN Gateway
connected to the PBX, such as those made by Vega/Sangoma. Please call us for confirmation if you plan to
connect one to the service.

SIP TRUNK INTEROPERABILITY (Compatible Platforms & Hardware)

Asterisk based platforms (Confirmed)

• IP Cortex
• FreePBX v2
• TrixBox
• SARK
• Asterisk

FreeSWITCH based platforms (Confirmed)

• FreePBX v3
• FreeSWITCH

Broadsoft based platforms (Confirmed)
Metaswitch based platforms (Confirmed)
PABX Manufacturers (Confirmed):

• Panasonic NCP500 and NS1000 (accredited)
• Mitel 3300 (MCD) (accredited)
• Mitel 5000
• Samsung OfficeServ 7000 series (accredited)
• Avaya IP Office
• LG iPECS
• NEC SL1100 and SV8100 (accredited in IP authentication mode)
• Gigaset PRO T300/500
• snomONE PBX
• Cisco UC320 and IOS PBX
• 3CX Phone System
• Grandstream PBX
• iQ PBX
• DrayTEK PBX
• Yeastar S-Series IPPBX (accredited)

SBC Manufacturers (Confirmed)
• OpenSIPS
• GENBAND
• SONUS
• Kamailio
• OpenSER
• ACME Packet
• Metaswitch
• Dialogic (IP authentication only)
Incompatible Platforms & Hardware (Cannot be used)
• Microsoft Lync
• Skype for business (currently under testing – SEE THE DEVELOPMENT SECTION)
• NEC XN120

SIP TRUNKS & NETWORK

SIP Trunks offer a reliable, cost-effective and feature rich replacement for ISDN Circuits and
analogue lines.

Our SIP Trunks are instantly scalable and can be provisioned in real-time.

We include our advanced fraud prevention service as standard.

Calls can be served to your SIP Trunk in HD (high-definition).

When making outbound calls through your IP PBX, that call travels through your Data Connection to reach our Call Routing platform, which then ‘hands-off’ the call to the upstream networks. When receiving a call from the PSTN network, the call reaches our Call Routing platform, which then forwards the call to your IP-PBX through the SIP Trunk(s).

Your inbound and outbound calls will follow bespoke call routing patterns, according to rules configured in our easy-to-use web portal. The web portal allows for easy initial configuration, as well as real-time changes to your office phone system if required. Because the Call Routing platform is hosted off-site, and accessible online at any time, you do not need to book an engineer if you wish to make any changes. Because our Technical support team is available 24/7, and conveniently based in our Birmingham offices, help is available if you have any difficulty with these changes.

• Our platform is intuitive and easy to configure. It’s been designed to be used by real customers without a background in Telecoms, not just engineers.

• Our SIP Trunking solutions are deployed from the same control panel as Hosted PBX features, giving you all the tools you need to build a highly advanced hybrid solution for your business. SIP Trunks and Hosted PBX functionality can be fused together, opening up possibilities that previously never existed.

• The platform functionality continues to evolve as the needs of our customers evolve. We’re constantly adding features that have been requested by our customers, and looking for new ways to make the lives of our customers easier.

• We use the same systems for provisioning as our customers. This means faster help when you need it, without the technical jargon.

We work with Tier 2 Telecommunications provider, which means we can operate our own Call Routing infrastructure and telephone number ranges. However, we do not maintain the National Infrastructure (commonly known as the PSTN network) used to deliver the calls themselves. Instead, we use the existing infrastructure (through multiple interconnects) managed and maintained by Tier 1 and 2 Telecommunications providers. All calls utilise the extensive British Telecom PSTN network, ensuring optimal call quality, reliability and scope.

Our SIP Trunks are RFC3261 compliant.

Network

DMV operates with World-Class carrier-neutral data centre partners Equinix and Telehouse along with direct private fibre only connections and membership into LINX and LONAP, the largest IXPs in the UK and around one the largest IXPs in the world, means we’re operating inside a network of interconnected switches at major data centres in the UK. This provides DMV SIP service with a professionally run, uncongested peering and transit fabric and low latency routes from all major global networks.

Network Resilience
• N+1 Redundant Private Fibre Interconnects
• Failover and Diverse Tier 1 only Transit Routing
• Proprietary Load-balanced Voice Gateway Array
• World Class Carrier Neutral Data Centre Partners (Equinix and Telehouse)
• Private Gateways
• Multiple IXP [Internet Exchange] Members
• 99.99% Platform Up-time
• Custom Firewall Protection

Hardware Protection
• N+1 Redundant Power Systems
• N+1 Redundant Core Systems
• Cooling and Temperature Management
• Fire detection and suppression to BS 5839, 6266, 5445, 5588, 5306 and 3115
• Load-balanced Gateway Array

Privacy and Security
• Call encryption ensures complete privacy for all on-net endpoints (includes secure conferencing)
• Encryption of all call recordings
• Custom user privileges
• Anti-Fraud suite
• Payment Card Industry Data Security Standards (PCI-DSS Data-Centre Co-Location)
• ISO 27001 Data-Centre Co-Location

Green Commitment
• NO PRINT policy | Paperless Workplace
• Data centre footprint powered by 100% renewable energy
• Data centre waste heat exchange system facility to reuse lost heat to distribute to third parties and also pre-cool the chilled water, thus reducing both chiller and dry air cooler power requirements
• Data centre ISO 14001 Environmental Standard and the Carbon Trust Standard

SIP Trunks – Quick Start Guide

SIP Trunking Overview The VoiceHost SIP Trunking solution provides substantially more functionality than traditional ISDN circuits, at a significantly lower cost. By replacing your ISDN circuits with SIP Trunks over a VoiceHost Data Connection, you are able to make and receive calls to/from anyone in the world, make your calls follow advanced routing patterns and instantly activate any additional functionality you require such as call recording, conferencing, voicemail and TPS. While an independently sourced engineer will be required to implement the SIP Trunk in the first instance, later configuration changes to the SIP Trunk are instantly activated from our self-service online portal 24/7. Is it right for my Business? If your business needs to make and receive calls, and is in an area with high speed broadband, then the VoiceHost SIP Trunking solution will not only save you money, but will also open up a host of additional functionality for your phone system.

Hybrid Cloud Solutions Create complex and resilient solutions by using our features to compliment your PBX via our SIP Trunking. Take advantage of the power of our platform to create secure and resilient deployments.

Elastic Cloud & Enterprise Features Our real-time platform empowers you to choose only the features you want.Telephone numbers from around the world and route calls to users via call groups, call queues, time profiles and interactive voice recordings (IVR) to name just the most popular. As the VoiceHost cloud is developed new features become instantly available for use without the need to upgrade software.

SIP Trunk Features
Inbound Call Features
• Geographic (01, 02) Numbers
• Non-Geographic (03, 08) Numbers
• Free-Phone (0800) Numbers
• International Numbers
• Inbound Caller ID, with/without Prefix
• Call Recording
• Call Groups
• Call Queues
• IVT/ Auto attendant Menus
• Time Profiles

Business Features
• Conferencing
• Music On Hold
• Company Voicemail
• Company Diverts
• Custom prompts
• Fax to Email
• Dial-through
• Automatic Failover
• Dynamically Scalable
• 24/7 Support

Outbound Call Features
• Call Recording
• Daily CDRs
• Outgoing Caller ID
• Anonymous Dial
• Telephone Preference Service screening
• Live Blacklist filtering
• IP Authentication

Fraud Prevention
• Daily Call Spent Limit
• Call Spend Limit Alerts
• UK Premium Rate Number Block
• IP Address Lock
• Proactive Fraud Monitoring
• Time based Restrictions
• Destination based Restrictions

While many businesses choose to use SIP Trunks rather than a full Hosted PBX solution, our control panel makes it fast and easy to add functionality from the Hosted PBX to your legacy IP-PBX. By using Hosted PBX features in conjunction with a SIP Trunking solution, you can greatly increase the capabilities of your IP-PBX at a significantly lower cost than purchasing additional equipment and line cards for it. Also, because our Hosted Control Panel allows you to integrate SIP Trunks and Hosted PBX Seats from a single login, you’re able to incorporate a mixture of the technologies across different sites. For example, you can easily add remote workers to your business, and benefit from free calls between the different sites.