Freeswitch

Freeswitch is an open source telephony platform utilizing a large number of external libraries, including Sofia-SIP for SIP support.

SIP features taken from [1]:

UDP, TCP, SCTP and TLS transports for full sip compliance.
SIP Session timers
RTP Timers
IPv6 Support
RFC3263 (SRV and NAPTR)
SRTP via SDES (works with polycom, snom and grandstream)
Blind SIP Registration
STUN Support
Jitter buffer
NAT Support
Distributed sip registrations
Late Codec Negotiation
Multiple sip registrations per user account.
Multitenancy - Multiple sip UAs
SIP Reinvites.
Can act as an SBC (session border controller)
Manage Presence
SIP/SIMPLE (can gateway to other chat protocols)
SIP Multicast Paging support for Linksys and Snom
Intercom/AutoAnswer support.
Call features like Call Hold(Re-INVITE), Blind Transfer(REFER), Call Forward(302) etc

FreeSwitch also has it's own interop list.