[Post is better viewed on the blog Website]
When I was a kid, I loved reading the Dr. Seuss books with the fun and educational rhymes. As an adult, I loved reading them to my own children when they were young. Some things never grow old.
It seems that it is the same with the quality of a phone call. From the early days of the switchboard to calls over the cellular network, the phrase “can you repeat that? I couldn’t hear you” has been commonly heard over the years. And so has the emphasis on improving voice quality accompanied the advances made in telephony communications.
Source: One Fish Two Fish Red Fish Blue Fish, by Dr. Seuss
This is no less true today with Voice over IP (VoIP) networks. Perhaps the greatest enemies of a quality call over VoIP networks are delay, jitter and packet loss. Network experts invest great efforts in fine tuning their networks so as to minimize the negative effects of these factors. In a recent post on this blog entitled “Voice is Coming to LTE”, Amir Zmora pointed out that in his experience in speaking with service providers that are already invested in VoLTE and interconnecting with other networks, it was clear that voice quality and QoE are their main concerns.
VoIP traffic may traverse wireline, WAN, cellular or wireless networks. Wireless traffic in particular is inherently inconsistent and the effects of delay, jitter and packet loss, if not handled, can seriously impair the quality of a call. Wireless networks were designed first and foremost for data applications. But in data focused applications, the most important thing is for the payload to arrive complete, how fast it gets there is less important. Thus, compensating measures can be implemented to guarantee that requirement. However, while these capabilities ensure the arrival of the packets, they also increase delay and jitter. And while delay for data applications is acceptable, in the case of VoIP the delay may be intolerable resulting in someone saying “I didn’t catch that, come again?” or just dropping the call altogether.
As most VoIP entities in the network (session border controllers, gateways, ATAs, IP Phones, mobile clients, etc.) were designed to handle wireline, and not wireless impairments, they have a hard time handling (without help) traffic emanating from wireless networks. To get around the problem, AudioCodes has implemented special built-in tools inside its Session Border Controllers. By placing the SBC with these tools between the wireless network and any other network (wireless, wireline, cellular, etc.), the impairments from the wireless network traffic can be managed and reduced dramatically to allow for an end-to-end quality call and prevent someone from saying, “I cannot hear your call. I cannot hear your call at all!”
Want more information?
Click here to download a White Paper in which you will learn how built-in SBC tools such as an adaptive jitter buffer, transcoding, redundancy, trans rating and quality-based routing can each play a major role in significantly enhancing QoE. Furthermore, when taken together and given the ability to fine tune and balance between the variables, they can be truly powerful.