A little linphone audio client. Only speex16 and ultra-minimal. More a hack than anything else.
Linphone for GNU/Linux does not work well for me and is way too big to debug (and to compile too). To hell with that bloat, let's make a very simple and basic SIP client to use with the linphone's servers (the initial release is small, around 1000 lines of code) (yes, one thousand) (linphone is I-don't-know-how-many lines of code, but way-too-much-for-what-it-does-so-bad).
To compile you need:
- speex (I used speex-1.2rc1) - libeXosip2 (I used libeXosip2-4.0.0) - libosip2 (I used libosip2-4.0.0) - alsa stuff
You also need to edit the Makefile.
Get rid of the osip libraries. They suck.
Handle SDP "correctly".
What about NAT traversal, hmm? (Another madness in this world.)
There is no security at all. SRTP is not handled. SIP is used without TLS (you can change that one very easily in the code I think) (but do you get some correct secure behavior, I don't know).
The code is so full of bugs that anything may happen. Don't trust this code. I repeat: DON'T TRUST THIS CODE! (Why release it then?) (For I can.)
Ah, and I've seen that linphone decides to intercept your calls. There is no direct connection between you and who you call. Linphone server is in between. Don't ask me why. Maybe a bug in my code, or maybe the bug is on their side. Or is it a bug? Maybe linphone spies on you.
So I repeat: DON'T BLOODY TRUST THIS CODE, USE AT YOU OWN RISK, BLAHBLAHBLAH.
SIP sucks. Bloated. Linphone is no better. Hey people in the VOIP shitty business, can't you buy yourselves decent brains? Assholes.
Fri, 06 Sep 2013 23:36:09 +0200
Last update: Fri, 06 Sep 2013 23:36:09 +0200