[VOIPSEC] Asterisk with SRTP and SIP over TLS

[Iñaki Baz Castillo]

El Viernes, 20 de Marzo de 2009, Klaus Darilion escribió:

> Although the SIP stack of Asterisk is not the best, from my point of
> view Digium is doing lots of work on the SIP stack to make it more
> stable and adding new features.

Yes, chan_sip.c already has near 20000 lines of code (a single file). It's 
really great.

It's also nice the propietary and limited SIP line monitorizing mechanism 
implemented in Asterisk, instead of RFC 4235 which you know very well :)
Due to this, some phones as Linksys already include an option "BLF type: 
Asterisk". It's so nice for SIP interoperability... thanks Digium.

And better if I don't mention cool Asterisk's SIP features as 
wrong-spiral-detection, registrations-limited-to-one-user, 
great-codec-negociation, first-record-SRV-support and 503-for-any-reason.


-- 
Iñaki Baz Castillo