Voice and data can dynamically share capacity, and the capacity can flex to the business needs. Perhaps the most visible role SIP is playing in the center today is on the trunking side. Alternatively, the recording server can connect to the LAN/Ethernet switch on the network that the VoIP communication is traversing and capture the data via port spanning (basically replicating the voice packets). A Session Border Controller (SBC) connects to the recording server, and a Session Recording Protocol (SIPREC) interface triggers capture of both the voice stream and the signaling messages from the SBC. SIP is increasingly the answer for quality monitoring. SIP can also be the source of information for screen pops (for example, the “address” in a SIP message can be used to look up the customer). A smartphone session could use SIP to click-to-dial from a mobile app. Because it’s a multimedia protocol, a session could be established for one media and cross to another, such as a chat session evolving into a phone call, or a phone call into a video call. For voice, it can be the call control for all calls, including IVR interactions. While SIP isn’t the only way to deliver functionality centers seek, it is a good way. It lets companies use their internet connections if they choose, or a dedicated connection if they prefer. SIP trunking uses what we typically think of as data connections to carry voice and other media. Some are even pushing only SIP at this point, clearly showing the migration from the traditional Public Switched Telephone Network (PSTN) is on. Solution vendors and carriers support SIP trunking. You’ll want to select from approved options that have been interoperability tested, or take on that burden yourselves. However, don’t assume that any SIP phone works with any SIP-based system. When systems are truly SIP compliant, buyers have choice and presumably find lower cost phones. A contact center technology solution provider may have their own SIP phones (true SIP or proprietary version), or you may be able to use a third party product from companies like Polycom, Yealink, Panasonic, Audiocodes, or Spectralink. Many phones are SIP-based, whether traditional “hard” phones with handsets, or “soft” phones that are software based and typically use a USB headset. This may sound like the best of both worlds to some, but a failure to comply with standards to others. In these cases, they are basically offering a proprietary protocol that is based on SIP but extends beyond it to do more. Some vendors have their own version of SIP to address issues such as encryption and security, or offer greater functionality. It can also provide security, acting as a firewall, and manage quality – two areas where SIP needs help outside the protocol. The SBC can govern both the signaling (SIP) and the media stream (e.g., RTP). A Session Border Controller (SBC) is often the termination for trunking and sits between the network and the IP-based PBX (phone system). Endpoints such as phones register with the proxy server or registrar. But SIP isn’t just about voice – it can manage sessions for a variety of media (e.g., voice, video, chat), supporting the “omni-channel” contact center.Ī SIP configuration typically involves a proxy server, registrar, and/or SIP server that helps with the address resolution, determining where to actually point a given address. SIP fits with “Voice over Internet Protocol” (VoIP), which is the way most voice communications works today, regardless of vendor or solution. It is one of many standards that have been defined over the years, and is competing with or replacing predecessors such as H.323. SIP is a protocol standard for initiating and managing “sessions” or connections between point A and point B.
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