VoIP telephony
Improving internet telephony
IP telephones, softphones, smartphones: VoIP telephony is possible via a range of different digital devices. However, the IP network is not designed for voice transmission. There is no fixed channel assignment or guaranteed transmission bandwidth, nor are the network and devices clock synchronized. VoIP devices must be capable of constantly adapting to varying network conditions. To still achieve an optimum call quality in the sending and receiving direction, developers need to perform more than just standard device testing – they also need to look at the impact of the network properties on VoIP telephony.
Testing and analysis
Delay, jitter and packet loss can all affect VoIP telephony. They have a huge impact on voice quality and speech intelligibility. This is why it’s so important to understand the causes and to test and compensate for the effects.
Unlike in circuit switched transmission, devices are not synchronized to each other in IP telephony. This is why good buffer and PLC management is key for seamless conversation.
This means that VoIP telephony can be optimized comprehensively and effectively only with advanced measurement procedures and simulations of the network environment.
Optimizing VoIP telephony at all levels
Our software and hardware solutions allow manufacturers of VoIP telephones to test and optimize the communication quality of their devices at all levels.
For this purpose, we offer:
- Realistic and reproducible simulation of acoustics and the network environment
- Automated tests in accordance with international standards (e.g. ETSI ES 202 738-740, TIA-920)
- Numerous manufacturer-specific measurements and standards
- Advanced measurement procedures such as Listening Effort (ABLE), 3QUEST, EQUEST, TOSQA, POLQA
- Decades of experience in VoIP telephony