Our recent tests showed that latest enterprise grade 802.11 AC access points are capable of delivering cumulative throughput of the order of up to 300 Mbps, with 100 concurrent users in real life situations. So what would you expect if we do a concurrent VoIP call test which needs a cumulative throughput of just 150 Mbps. This should be a cake walk for the access point, right??
The reality may be far away from expectation!!! Throughput may not be a good indicator of access point performance when the application is VoIP.
Let’s see what’s so special with VoIP!
Voice over Internet Protocol (VoIP) is a technology that allows you to make voice/telephone calls using a broadband Internet connection instead of a regular phone line, which has been the method used for the last hundred years or so.
Early VoIP deployments started in 2004. VoIP picked up for the primary reason that it is low cost to the end user as it rides on existing multi-purpose IP networks. Businesses are migrating from traditional telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new (PBX) lines installed internationally were VoIP!
At a high level, what is needed to implement VoIP protocol is similar to traditional telephony. It involves signaling, channel setup, digitization of the analog voice signals, and encoding. They transport audio streams using special media delivery protocols that encode audio with audio codecs. Various codecs exist that optimize the media stream based on application requirements and network bandwidth
VoIP as a protocol has multiple implementations. In the early 2000s, while I was working on implementing VoIP over 3G Phone, H.323 standard was used. Then technology moved on and players like Skype implemented their own methods. Among the standard protocols today, SIP is the most popular method of implementing VoIP.
Session Initiation Protocol (SIP) is a text-based, application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints.
- Does necessary signaling to locate and invite the target endpoint based on address
- Determines the lowest level of common services between the endpoints through Session Description Protocol (SDP) and then establishes a two-way voice path via Real-time Transport Protocol (RTP).
- Handles the termination of the call.