Co-founder of AvaritCall
Seyit Ahmet Kutuzman
Works on artificial intelligence, telephony infrastructure and scalable software systems.
About
Seyit Ahmet Kutuzman is a co-founder of AvaritCall and leads the company's software and infrastructure. The focus is the engineering that turns AI into reliable systems that carry real calls every day without interruption.
At the centre of this work is the invisible layer behind voice AI: service deployment and continuous delivery, server infrastructure, integration between services, and connecting large language models to business processes.
A graduate of Istanbul Technical University (ITU) with training in artificial intelligence and machine learning, Seyit has worked at the Turkish Ministry of Trade and has experience in enterprise software development in the financial sector. Earlier, Seyit founded the independent game studio Linden Games and built games for a range of clients, which is where an interest in real-time systems began.
On the blog, Seyit writes about the infrastructure side of AI call center automation: carrying voice over SIP and RTP, codecs and audio quality, securing phone audio, integrations and setup.
Areas of expertise
- Software development
- Artificial intelligence and machine learning
- Voice AI infrastructure
- Server and cloud infrastructure
- Continuous integration and delivery (CI/CD)
- Large language models (LLM)
- Real-time systems
- Game development
Articles
14 articles
Not just a voice: AvaritCall talks to your CRM, calendar and systems
The keypad menu is dead: from 'Press 1 for...' to an AI that actually talks
How RTP packetization changes bandwidth and delay
SIP, SDP and RTP: from codec negotiation to working audio
How jitter and packet loss affect voice AI
8, 16 and 48 kHz: a reliable STT path from μ-law to PCM
G.711, G.722 and Opus: choosing a call audio codec
SIP 403, 408, 486 and 503: investigating call failures
MOS, PESQ and POLQA: measuring telephone speech quality
Telephone echo and AEC: why your own voice comes back
DTMF interoperability: in-band tones, RFC 4733 and SIP INFO
TLS and SRTP: what is protected in a telephone call?
WebRTC, SIP and PSTN: choosing a voice AI call path
