PJSIP is a free and open source multimedia communication library
written in C language implementing standard based protocols such
as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol
(SIP) with rich multimedia framework and NAT traversal functionality
into high level API that is portable and suitable for almost any
type of systems ranging from desktops, embedded systems, to mobile
handsets.
WWW: https://www.pjsip.org/
===> The following configuration options are available for pjsip-extsrtp-2.9_4:
AMR=off: AMR 3GPP speech codec support (opencore)
DEBUG=off: Build with debugging support
FFMPEG=off: FFmpeg support (WMA, AIFF, AC3, APE...)
G711=on: G.711 codec support
G722=on: G.722 codec support
G7221=on: G.722.1 codec support
GSM=on: GSM codec support
ILBC=on: iLBC codec support
IPV6=on: IPv6 protocol support
L16=on: Linear/L16 codec support
OPENH264=off: OpenH264 support
PJSUA=off: Command line SIP agent
RESAMPLE=off: Enable resampling implementations
RESAMPLEDLL=off: Build libresample as shared library
SAMPLERATE=off: Sample rate conversion support
SDL=off: Simple Direct Media Layer support
SHARED=on: Build shared libraries (other ports may depend on this)
SOUND=off: Sound (audio) support
SPEEX=on: Speex audio format support
SPEEXAEC=on: Speex Acoustic Echo Canceller/AEC
V4L=off: Video4Linux2 support
VIDEO=off: Video support
WEBRTC=off: Build linwebrtc
===> Use 'make config' to modify these settings
Add a slave port to net/pjsip to force installing pjsip with external
SRTP library.
Make the www/asterisk13 depend on this slave port when both SRTP
and PJSIP options in it are enabled, this allows enabling SRTP
support in asterisk13 without the need to manually reconfigure other
ports.
Reported by: mat@ and a few others