SIP trunking is an internet connection established using SIP between a business IP-PBX to Voice over IP (VoIP) services provided by an Internet Telephony Service Provider (ITSP). Phone calls can originate as an IP call or an internal POTS call that is converted to IP by an analog telephone adapter (ATA) or media gateway and then transmitted over the SIP trunk to the ITSP. Unlike the physical line and switching required with PSTN service, the IP-PBX uses virtual phone lines to access ITSP VoIP services over a broadband internet connection. Trunk bandwidth must be adequate to handle the day-to-day voice, data, and video usage. Business or telephony QoS will be affected when the connection to the ITSP is overloaded and packet latency and loss increase.
SIP trunking allows businesses to utilize lower cost VoIP services and calling plans provided by an ITSP. If multiple business locations want to access VoIP services, each location can utilize a separate SIP trunk to the ITSP from its local IP-PBX or the remote locations can direct calls over an internet connection to the local IP-PBX and access the SIP trunk. Many IP-PBX systems support dedicated internet connections between multiple locations. Thus businesses could purchase larger bandwidth from one location to take advantage of bundling discounts with the ITSP and share or allocate the bandwidth amongst the individual offices. This also permits offices to use the internet to connect through another location to access the local PSTN system and use the local call service rates.
VOCAL’s embedded libraries include a complete range of ETSI / ITU / IEEE compliant algorithms, in addition to many other standard and proprietary algorithms. Our SIP source code is optimized for execution on ANSI C and leading DSP architectures from TI, ADI, AMD, Intel, ARM, MIPS, and other vendors. The SIP software libraries are modular and can be executed as a single task under a variety of operating systems or standalone with its own microkernel.