Re: RFA: rtc.debian.org
>>>>> "Jonas" == Jonas Smedegaard <dr@jones.dk> writes:
Jonas> I also run Asterisk at a few small networks, and experienced
Jonas> similar failure connecting with rtc.d.o in the past - but I
Jonas> didn't try very hard back then.
Jonas> I sincerely hope these renewed efforts can help improve the
Jonas> experience.
Jonas> No doubt helps also that the protocols and implementations
Jonas> have since matured some.
I played around with asterisk and rtc in browsers.
The issue I ran into was that libpjsip had several compile-time
constants that produced buffers too small for some of the SIP messages
that webrtc uses.
When you're using audio, video, DTLS, and ICE, your sip messages kind of
get big.
I never tried with rtc.debian.org but I did get things working with
chrome, firefox and asterisk.
Unfortunately, I think that increasing the compile-time constants may
break the pjsip API, which may be why Debian doesn't do it.
(It may have sense been done; this was a while ago)
Honestly I think an organized ABI breakage would be worth support for
webrtc that works with real browsers in asterisk.
If I weren't being DPL at the moment, I totally would have volunteered
for this myself.
I'd certainly be open to being roped into a couple hours helping at
debcamp or something.
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