Duplex voice calls in Sideband

[Skretch1974]

Hi. There is a proposal to add to the Sideband application the ability to make duplex voice calls with the same list of codecs and bandwidth control by their speed, as it is now implemented in the
Reticulum-meshchat. Our team is actively using and testing voice calls and it works just fine even with the use of mobile internet on a 2G (edge) connection, which currently has the best coverage in remote areas away from towns and cities. It is also now possible to bridge WiFi radio bridges over long distances and create separate isolated Lan networks through which voice calls can be made.
I also suggest to consider adding another transport interface using a regular telephone audio channel as a transport, as implemented in the jackpair project - https://github.com/gegel/jackpair.
Here through any analog audio channel a digital duplex channel is created, through which all communication on low-bitrate codecs for voice calls or data and text messages is carried out.

[resiliencetheatre]

I really love to participate this without using WebRTC. Maybe I could check how current implementation pipes packets between entities.

[markqvist]

For a while, I've been working on a protocol tentatively dubbed "RMV", Reticulum Voice Mesh. You could think of this as the real-time audio equivalent of LXMF. The reason that I have not yet implemented real-time audio in Sideband is, that when I eventually do, I want it to be based on a flexible, open and extensible protocol, like the messaging is currently with LXMF.

Some of the primary design for RVM are:

• Work natively over Reticulum
• Provide a secure, universal and flexible Reticulum-native protocol for real-time audio
• Provide both simplex and duplex call functionality
• Provide group/conference call functionality
• Provide functionality for broadcast audio (think public radio broadcasts over ISM and license-free spectrum)
• Provide a variety of audio encodings based on open and free standards
• Be easy to integrate with legacy VoIP/POTS phone systems

I have no idea how far out an actual release of this is though.

[Skretch1974]

Hello. There are some peculiarities in realization of voice transmission in low bandwidth networks. A few years ago one author under the pseudonym Gegel independently implemented a unique at that time project on voice transmission through TOR networks, having developed his own modernized codec based on Melpe codec, able to work in networks with delays more than 3 seconds, and also developed his own protocol over tcp for voice transmission. You can read more about his research with the help of an interpreter at https://habr.com/ru/articles/448856/.
He has also implemented a software audio modem to establish a digital duplex communication channel over a standard telephone audio channel, for example, one can make an audio call on Skype and run a digital data channel for both voice and text messages over this connection. This is potentially another type of transport layer for the reticulum project, allowing, for example, from an area where there is no internet or digital radio modem coverage but a standard telephone wireline connection, to implement a digital gateway to the reticulum.
If you, as a developer, are interested to see Mr. Hegel's developments, I have kept a complete archive of his files and developments. https://web.archive.org/web/20160113114629/http://torfone.org

[markqvist]

That is interesting work. Thanks for the references, and for the archival effort! I will look into it.