VoIP Gate (SIP Trunk)

Service Manual
VoIP Gate (SIP Trunk)
Description
VoIP Gate connects your IP-based communication’s
system (e.g. an IP PBX or UC Server) to the public telephone network (PSTN) using the future proof Session
Initiation Protocol (SIP Trunking). VoIP Gate utilizes the
existing data network, which reduces costs and promotes a basis for transporting a variety of communication services on top of a data network . Due to prioritization of voice communication on Swisscom data
network, the quality of speech is guaranteed. In addition, the state-of-the-art voice over IP network fulfils
the highest security and reliability requirements. SIP
Trunking is an ideal solution for a company operating
throughout Switzerland which would like to centralise
and simplify its communications structure and therefore reducing the total cost of ownership in acquiring
next generation communication systems. Your complete number plan can be distributed over one VoIP
Gate or several VoIP Gates. VoIP Gate offers the basic
telephony services plus a range of additional service
options in order fulfil the requirement of customized
solutions ranging from complex redundancy architecture to flexible emergency number routing. Several
profiles with VoIP Gate are available It is possible to
order a VoIP Gate with an integrated IP access (limited
services) or to connect the VoIP Gate to a traditional
TDM PBX with ISDN PRI interface. All communication
between company sites, connected through VoIP Gate
is free of charge. All your existing number ranges can
be ported and used with VoIP Gate. You can analyse
your calls through e-Invoice available in Extranet.
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Profiles
Default profile of VoIP Gate uses SIP Trunking and does not include IP Access (LAN-I service). A number of concurrent
calls is ordered. It is possible to obtain the following profiles
VoIP Gate Connect
This SIP Trunking profile includes a dedicated IP Access over MPLS with the necessary
bandwidth and correct Class of Service throughout the network.
ISDN Media Gateway
A Gateway connects a traditional TDM PBX with ISDN Primary Rate Interface (PRI) to VoIP
Gate. The conversion of ISDN protocol to VoIP (H.248) occurs in GW
Basic Telephony Characteristics
VoIP Gate offers all the main characteristics of traditional TDM based telephony
Speech quality
The subjective perception of speech quality is comparable with the voice quality of traditional SDN lines with the condition that the G.711 a-Law encoding is used and voice packets are prioritized over data packets throughout the whole communication path
Number display
The number of the calling party is displayed in the phone’s display.
Direct dial
External callers can dial an internal extension directly.
Touch tone dialling
Numbers selected can be sent through the following procedures, which are negotiated
between the voice system.& VoIP Gate once the call is established: in-band traffic G.711
(RTP Traffic) or out of band RTP Traffic, H.245 or RFC 4733
Fax
Integrated fax systems behind all certified voice systems with VoIP Gate are supported in
both G.711 in-band or T.38 (limited systems). Analog fax converters are available
Standard emergency
number routing
Location identification for emergency number routing can be defined permanently for
each block of phone numbers per Trunk (maximum three location per trunk)
Displaying a different
number
Outbound calls can be sent out with a different presentation number (sent by the voice
system) e.g. a 0800x number. The billing will be done on the main number
Service options
Following service options are offered Your customer adviser can inform you of the current prices
Call Redundancy
Upon the architecture of the communication systems, VoIP Gate offers several
redundancy options. The redundancy options cover failover mechanism for incoming calls from one primary communication system to a secondary system or
further systems based on hunt, round-robin or other predefined algorithms. VoIP
Gate can accept outgoing calls from several active communication systems (active-active), which may do load balancing as well. Commercially, only one SIP
Trunk with a certain number of channels is ordered.
In case of two or more active SIP Trunks, incoming call distribution can be
achieved between several communication systems (based on a defined priority
list or defined percentage of call distribution). In case of failure of the first SIP
Trunk (or communication system) or an call overflow, a failover will occur automatically to the second SIP Trunk. Different overflow mechanism can be defined
for each number range.
The redundancy on the IP layer (Layer 3) shall be considered during the design of
IP Access (LAN-I).
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Service options
On-demand activation
On-demand activation allows a company to have a switchover plan of VoIP Gate phone
numbers to a predefined set of numbers (mobiles or landline). In case of an emergency
failure or planned maintenance of voice systems, this service can be activated through the
helpline or extranet The predefined switchover plan can be changed without extra cost
over extranet. Monthly fee applies.
Individual emergency
number routing
It is possible to assign freely any individual numbers or block of numbers from your entire
number plan to any physical location for a single VoIP Gate (SIP Trunk) or several VoIP
Gates. If the location management is done by the customer himself through VoIP Gate
Selfcare, this option is free of charge. If the location data is handed over to Swisscom one
time- and monthly fee applies.
IP Redundancy (SDT1)
(VoIP Gate Connect)
Downtime protection for an IP-Access (SDT1). (valid only for VoIP Gate Connect).
Configuration Adjustment
The adjustment of configuration are part of the service options and do not cost. All new changes in basic profile e.g.
“more concurrent calls” or new service options will be offered by Swisscom Sales advisor
Swisscom Extranet
Changes on the predefined switch-over plan of the “On-Demand-Activation” service
Swisscom Helpline
All changes for basic service and options can be done through helpline
Customization
It is possible to make customization on SIP Trunking solutions through Swisscom based on customer’s needs. One time
charges for engineering services will apply. Swisscom offers a range of possibilities such as implementing “Private
Number Plans” on VoIP Gate for customers with a mixture of VoIP products, complex redundancy, or design-in of an
enterprise session border controller (SBC)
Moving of premises
When a physical location behind a VoIP Gate is subject to a move. A new VoIP Gate will be offered. The customer is responsible to order the IP Access.
When a move of a location behind a centralized VoIP Gate (SIP Trunk) is occurring (virtual move), the customer is
obliged to inform Swisscom about the new emergency number plan. When an ISDN Media Gateway profile is used, the
old gateway will to be removed by Swisscom.
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Compatibility list for the different types of private branch exchange (IP PBX) or communication servers
The following voice systems (several software versions) are certified by Swisscom against the VoIP Gate (SIP Trunk)
Aastra IntelliGate (150/300/2025/2045/2065)
Aastra 400 Serie (415/430/470)
Aastra MX-ONE
Alcatel-Lucent OmniPCX Enterprise (OXE)
Asterisk
Audiocodes Mediant SBC (only with Lync 2013 + Audiocode Mediapack Fax Gateways)
Avaya (Ex-Nortel) Communication Server 1000
Avaya Aura (Session Manager & Communication Manager)
Cisco Unified Communication Manager (CUCM + CUBE)
Dialogic SR-140 Fax Server
Kofax Communication Server (KCS FoIP)
Microsoft Office Communications Server (OCS) 2007 R2
Microsoft Lync Server 2010 (certified by Microsoft
Microsoft Lync Server 2013 (certified by Microsoft)
Oracle Acme Packet 3820/4500 Session Border Controller (Swisscom Managed SBC)
Unify Hipath 4000 / OpenScape 4000
Unify OpenScape Voice
Unify OpenScape Business
More upon request
Remarks
Upon the commissioning of VoIP Gate, the setting of the voice systems must be correctly configured by the customer’s
system administrator. This configuration can be ordered as an engineering service from Swisscom
In order to achieve high speech quality, the customer is obliged to prioritize the VoIP packets through the entire communication system and network
The ISDN Media Gateway, in case needed, is provided by Swisscom
VoIP Gate can be combined with any other VoIP Gate and VoIP Phone (IP Centrex) connections
VoIP Gate can support both G.711 a-law encoding and G.729. Although G.711 is recommended and used by default
The necessary bandwidth on your IP access (Swisscom LAN-I service) & Class of Service (platinum) will be calculated
and communicated to the customer by Swisscom. The IP Access is included in the VoIP Gate Connect profile.
The information in this document does not constitute a binding offer. It is subject to revision at any time.

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