The term “VoIP Trunking” usually refers to a trunk connection between a customer VoIP Network and the VoIP service provider who provides the connection to the Public Voice Networks such as the PSTN and Mobile networks. The VoIP Trunk is also more often called a SIP trunk after the protocol used to setup, manage and clear calls. This protocol is called Session Initiation Protocol.
SIP Trunks are quickly replacing traditional voice trunks that mainly utilise ISDN (Integrated Services Digital Networks) technology. The SIP Trunk is used to connect the customer to the public network and the service provider will port a subscriber’s number or numbers on to it’s own telephone exchange. The service provider will invariably use an Internet Protocol connection to carry the SIP exchanges back to its network, and it is essential that Quality of Service is maintained throughout the transfer of customer VoIP connections. This is normally achieved by using a leased line connection, or more recently an MPLS (Multi Protocol Labeled Switching) or VPLS (Virtual Private LAN Service) connectivity.
SIP Trunks come in a variety of guises such as:
A managed service where the provider supplies Customer Premises Equipment and is then able to monitor the trunk connection and provide guarantees about the reliability of the service.
An unmanaged service which only provides a simple, basic phone service.
A SIP connection between two applications
Traditional legacy trunking comprised T1 or E1 ISDN lines which were channelized, with each separate channel carrying a single phone connection. SIP can use almost any IP capable trunk, and in the case of a T1 or E1 line can operate with the line not having any channelization. The line or trunk can simply be thought of as a pipe providing a certain amount of bandwidth over which SIP connection can be carried within IP packets, with as many separate conversations as the bandwidth will allow.
The number of simultaneous calls that can be carried across a SIP Trunk will be dependent on the amount of bandwidth that has been made available and the voice codec chosen for the connections. Each call will consume a certain amount of bandwidth determined by the voice codec, which is often G.711 or G.729, although other voice codecs can be used. A call using a G.711 codec will need approximately 95Kbps per simultaneous call, whereas the G.729 codec will only require around 40Kbps. G.729 SIP Trunks are the most common and the most efficient due to the lower bandwidth requirements and also the fact that this codec provides good compression with only a small drop in call quality.
The method used to deploy a SIP Trunk will be determined mainly by the customer’s PBX (Private Branch Exchange). Most IP-PBXs are configured to set up a native SIP trunk without the need for any protocol conversion, whereas some older PBXs may require an interface in the form of a gateway usually supplied by the SIP Provider. In some cases the gateway will have to convert SIP messages to messages compatible with TDM (Time Division Multiplexing) standards such as ISDN. This is referred to as SIP to PSTN internetworking and can take a number of forms, two of which are complete protocol internetworking and protocol encapsulation or SIP-T (SIP Telephony). SIP messages are converted to the equivalent ISDN messages outbound and from ISDN to SIP in the reverse direction.
There are two distinct components parts to a SIP voice call:
The media in the form of a codec which can be different on the SIP Trunk and in the PSTN to which the call is connecting. For example, G.729 on the Trunk and traditionally G.711 with the PSTN TDM systems. Translation between these codecs will be performed in the gateway which will have an IP port on the on the customer side that is bridged to a TDM trunk on the provider side.
The signaling involves translating the SIP signaling messages to one of a number of PSTN signaling standards, predominantly ISUP (ISDN User Part) of a common channel signaling system known as Signaling System 7 or SS7.
Source by David W Christie