SIP trunking connects IP telephony with traditional PSTNs (Public Switched Telephone Networks). It enables enterprises to communicate over IP not only within the company, but also to outside PSTNs, thus they can make full use of their installed PBXs (Private Brand eXchanges).
SIP trunking delivers several benefits, such as the following:
It eliminates costly ISDN BRIs and PRIs.
There is no need to invest in PSTN gateways and additional line cards as the enterprise grows.
Edge devices offer a low-investment path in adding new lines because theyare less expensive per line than the corresponding PSTN gateway.
It permits optimal utilization of bandwidth by delivering both data and voice in the same connection.
It gives maximum flexibility in dimensioning and usage of lines because capacity is not purchased in bundles of 23 (T1) or 30 (E1) lines.
It provides flexible termination of calls to preferred providers; calls to anywhere worldwide can be made for the cost of a local one.
Redundancy with multiple service providers and links is available.
Interface requirements currently differ significantly between SIP serviceproviders, although progress is being made to standardize the enterprise/SIPservice provider interface in standards bodies such as the SIP Forum.
The SIP trunking interface provides the following customization options when sending SIP requests:
The ability to send the P-Preferred-Identity (PPI) SIP Header field, rather than the P-Asserted-Identity (PAI) SIP header field.
The ability to send the domain name, rather than the IP address, in the host part of the SIP From Header field.
The ability to send the domain name of "anonymous.invalid", rather than the IP address, if the caller has Calling Line Identity Presentation Restricted (CLIR) active.
The ability to send the SIP From and PPI header fields with the identity of the transferring/forwarding party, rather than the calling party, when a call is transferred or forwarded.
The ability to always send SIP reINVITE requests with SDP.
These options are primarily relevant for SIP trunking to service providers that use a non-standard SIP interface—for example, Italtel. Each can be enabled and disabled via Telephony Control Server Assistant or the CLI. Telephony Control Server Assistant also provides the ability to use endpoint templates that automatically assign these SIP attributes to SIP trunking endpoints without having to manually assign each attribute individually.
Parent topic: Connectivity
SIP Trunking - CLIP (Calling Line Identification Presentation)
CLIP (Calling Line Identification Presentation) provides the called party with the identity of the calling party. The identity consists of name and telephone number of the calling party. The Telephony Control Server is able to send identity information for the CLIP feature to a SP (Service Provider).
SIP Trunking - CLIR (Calling Line Identification Presentation Restriction)
Calling Line Identification Presentation Restriction (CLIR) prevents the called party from receiving the identity of the calling party.
SIP Trunking - COLP (Connected Line Identification Presentation)
COLP (Connected Line Identification Presentation) provides the calling party with the identity of the connected party. The identity consists of name (optional) and number of the connected party.
SIP Trunking - COLR (Connected Line Identification Presentation Restriction)
COLR (Connected Line Identification Presentation Restriction) prevents the called party from divulging identity information back to the calling party.
SIP Trunking - Call Hold, Retrieve and Alternate
An established call may be put on hold so that the holding party is able to place another call. Meanwhile the held party may receive MOH (Music on Hold). This music on hold may be provided by either the holding party, or by a media server on behalf of the holding party or locally by the held party.
SIP Trunking - Call Transfer
Call transfer is a call re-arrangement of an existing call in which one party is replaced with another party. Initially, the transferor is in an active call with the transferee. Then the transferor initiates a call transfer to the transfer target. Finally the transfer target gets connected to the transferee.
Attended Call Transfer
Attended Call Transfer allows the user to transfer the call by going on-hook while the destination is ringing; also known as unscreened transfer. The party initiating the transfer receives an audible indication (ringback tone).
Blind Call Transfer
When a user transfers a call to another user without telling the caller that the call is being transferred. The party initiating the blind transfer does not hear an audible indication (ringback tone).
Semi-Attended Call Transfer
Semi-attended Call Transfer designates the following call transfer scenario: Phone A calls Phone B; Phone B answers, and decides to transfer the call to Phone C; Phone B completes the transfer while phone C is still ringing.
Call Pick-up allows subscribers to answer any ringing or camped-on station within the business group.
Call Diversion is the change of the destination of a call.
MWI (Message Waiting Indication)
MWI (Message Waiting Indication) is an indication that is rendered on the phone, to inform the user that a message is waiting. This indication involves typically a display indication, an acoustic indication or a lamp on the phone.
CCBS (Call Completion to Busy Subscriber) and CCNR (Call Completion to No Replay)
The CCBS (Call Completion to Busy Subscriber) and CCNR (Call Completion on No Reply) features allow a calling subscriber to be automatically connected to a busy or no reply called subscriber when that subscriber becomes available..
3PCC (Third Party Call Control)
3PCC (Third Party Call Control) can be used by Lotus Sametime Unified Telephony applications to generate and manipulate calls. These 3PCC generated calls may be calls routed via a SIP Service Provider.