This VoIP call center guide is a collection of resources on VoIP in the call center regardless of where you are in the learning or buying process. If you’re new to VoIP, we suggest you start at the beginning with our VoIP learning center covering fundamental issues.
VoIP in the call center guide
If you’re shopping for technology, skip ahead to the sections on evaluating VoIP vendors and case studies on using VoIP in the call center. If you’re currently managing VoIP in a call center, read the section on maintaining VoIP. In each section, we’ll logically guide you through our resources to maximize your learning experience.
1. First things first… VoIP defined
VoIP (Voice over IP – that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN).
2. Let’s look at the difference between VoIP and voice and data convergence
VoIP specifically refers to sending voice traffic over an IP (Internet Protocol) network and does not integrate data applications. Voice and data convergence refers to sending both voice and data (such as LAN traffic) over any data network (typically an IP network, frame relay network or ATM network).
IP telephony refers to any “telephone” type service carried over IP and includes voice, fax, video and even text messaging. VoIP involves only voice transmission.
H.323 is a standard approved by the International Telecommunication Union (ITU) in 1996 to promote compatibility in videoconference transmissions over IP networks. H.323 was originally promoted as a way to provide consistency in audio, video and data packet transmissions in the event that a local area network (LAN) did not provide guaranteed service quality (QoS).
Although it was doubtful at first whether manufacturers would adopt H.323, it is now considered to be the standard for interoperability in audio, video and data transmissions as well as Internet phone and Voice over IP (VoIP) because it addresses call control and management for both point-to-point and multipoint conferences as well as gateway administration of media traffic, bandwidth and user participation.
SIP (Session Initiation Protocol) is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming and virtual reality. Like HTTP or SMTP, SIP works in the Application layer of the Open Systems Interconnection (OSI) communications model. The Application layer is the level responsible for ensuring that communication is possible. SIP can establish multimedia sessions or Internet telephony calls, and modify or terminate them.
H.323 and SIP are often compared and do compete with each other for VoIP services. H.323 has been the early leader in this market so it is very popular, but SIP is arguably becoming the de facto standard for VoIP.