Hi! One of the main reasons I'm not yet using gnunet-conversation as my main tool for real-time voice communication is it's lack of integration with the POTS.
There are two aspects to this: * Using FXS interfaces on embedded devices (VoIP routers) to connect old-school phones and routing the calls via gnunet. * Tunneling calls to traditional POTS via gnunet-conversation, like SkypeOut and such. I'm personally more interested in the first application, and look forward to hints or opinions on implementing this. I like analog phones and simple embedded devices as the amount of code and complexity is very limited, there aren't millions of layers of different libraries with tons of possible vulnerabilities... I imagine that GNS could be used to e.g. store e164.arpa records, thus I could easily have my local numerical phone-book and use my analog phone to call friends on gnunet-conversation To handle the 'connect-with-the-POTS' case, we will need a way to transport a POTS-number with the call, so a gnunet-conversation-2-POTS gateway can know which number on the POTS one wanted to call and in case of an incoming call, it'd be nice to signal the caller-id of the caller as seen by the gateway. I saw there is the LINE parameter, and I wonder if it is suitable to deliver E.164 numbers (which can be long) for outgoing calls to the POTS. What is the intented of the LINE field? (the description in conversation.conf makes me think that it could be useful for embedded devices with 2 or more FXS interfaces) Signinalling the callerid will need something similar, but supposedly abusing the same field for that would not be such a good idea. In practice, OPUS is hardly supported in any of the traditional VoIP software. I found this patch for Asterisk https://github.com/meetecho/asterisk-opus Writing an asterisk channel driver for gnunet-conversation seems to be the most feasible and generic approach to solve both the above. It'd obviously be more elegant to have proper OPUS support in Asterisk and let it handle pass-through and recoding when really needed... If that really won't work, gstreamer can handle the recoding to slin16 inside the gnunet-conversation channel driver... I'm willing to put some time and effort into drafting/prototyping. I guess I do have quite some experience with hacking asterisk and other SIP/VoIP stuff. Cheers Daniel _______________________________________________ GNUnet-developers mailing list [email protected] https://lists.gnu.org/mailman/listinfo/gnunet-developers
