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.

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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.

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.

VoIP calls encoded with SILK: from RTP to WAV

SILK is a codec defined by Skype, but can be found in many VoIP clients, like CSipSimple . It comes in different flavours (sample rates and frame sizes), from narrowband (8 KHz) to wideband (24 KHz). Since Wireshark doesn't allow you to decode an RTP stream carrying SILK frames, I was curious to find a programmatic way to do it. In fact, this has also allowed to me to earn a "tumbleweed" badge in stackoverflow . You may argue that a Wireshark plugin would be the right solution, but that's probably for another day. Initially I thought it was sufficient to read the specification for RTP payload when using SILK ; the truth is that I had to reverse engineer a solution by looking at SILK SDK's test vectors. There, I discovered that a file containing SILK audio doesn't have the file header indicated in the IETF draft ("!#SILK"), but a slightly different one ("!#SILK_V3"). More importantly, each encoded frame is not preced...

Extracting Opus from a pcap file into an audible wav

From time to time I need to verify that the audio inside a trace is as expected. Not much in terms of quality, but more often content and duration. A few years ago I wrote a small program to transform a pcap into a wav file - the codec in use was SILK. These days I'm dealing with Opus , and I have to say things are greatly simplified, in particular if you consider opus-tools , a set of utilities to handle opus files and traces. One of those tools, opusrtp , can do live captures and write the interpreted payload into a .opus file. Still, what I needed was to achieve the same result but from a pcap already existing, i.e. "offline". So I come up with a small - quite shamlessly copy&pasted - patch to opusrtc, which is now in this fork . Once you have a pcap with an RTP stream with opus (say in input.pcap ) you can retrieve the .opus equivalent (in rtpdump.opus ) with: ./opusrtp --extract input.pcap Then you can generate an audible wav file with: ./opusd...

Giacomo Vacca

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