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About

I’m Giacomo Vacca. I work on real-time communications over the Internet: WebRTC, SIP/SDP, NAT traversal (ICE/STUN/TURN), RTP/media pipelines, and the tooling needed to debug them in production.

This blog is a place where I write down things I’ve learned while building and troubleshooting RTC systems: practical explanations, packet-level investigations, and small notes that I want to be able to find later.

If you are new, start here:

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Topics

What you will find

WebRTC connectivity and NAT traversal (ICE/TURN)

SIP/SDP behavior and negotiation details

RTP/media analysis from PCAPs, codecs, and trace dissection

Occasional off-topic posts about running

Contact

If you want to reach me, the simplest way is via the comments, or through:

- giavac at X

- Linkedin

- github


Disclaimer

The views and opinions expressed on this site are solely my own and do not represent those of any current or past employer, client, company, or individual.

All content is provided for informational purposes only. Any code snippets, configuration examples, or references to software products are provided “as is”, without warranty of any kind. You are responsible for evaluating and testing anything before using it in your own systems.

You are welcome to share and reuse this content provided that you include a clear link back to the original post.

Comments are welcome. Please keep them respectful and on-topic. I may moderate or remove spam, abusive content, or anything that is clearly unrelated.


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About ICE negotiation

Disclaimer: I wrote this article on March 2022 while working with Subspace, and the original link is here:  https://subspace.com/resources/ice-negotiation . This post in my personal blog is a way to ensure it doesn't get lost. There is nothing service-specific in it, I've made only minor edits and I hope it can be a good technical reference on the topic. WebRTC is a set of protocols that allow applications, typically running on Web browsers, to exchange media (audio, video, data) with other entities. Before media can flow, however, the WebRTC entities need to discover what type of connection is possible, and among the possible connections, what’s the best to be used. This needs to happen as fast as possible, so that users can perceive the service as instantaneous as possible. WebRTC includes protocols like STUN and TURN that are designed to facilitate the establishment of connections when a direct connection is not possible. The typical case is a computer inside a home or o...
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Troubleshooting TURN

  WebRTC applications use the ICE negotiation to discovery the best way to communicate with a remote party. I t dynamically finds a pair of candidates (IP address, port and transport, also known as “transport address”) suitable for exchanging media and data. The most important aspect of this is “dynamically”: a local and a remote transport address are found based on the network conditions at the time of establishing a session. For example, a WebRTC client that normally uses a server reflexive transport address to communicate with an SFU. when running inside the home office, may use a relay transport address over TCP when running inside an office network which limits remote UDP targets. The same configuration (defined as “iceServers” when creating an RTCPeerConnection will work in both cases, producing different outcomes.
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