| Name : baresip
| |
| Version : 3.11.0
| Vendor : obs://build_opensuse_org/network:telephony
|
| Release : lp155.16.3
| Date : 2024-09-10 05:22:49
|
| Group : Productivity/Telephony/SIP/Clients
| Source RPM : baresip-3.11.0-lp155.16.3.src.rpm
|
| Size : 1.78 MB
| |
| Packager : https://www_suse_com/
| |
| Summary : Modular SIP useragent
|
Description :
A modular SIP user-agent
with support for audio and video, and many IETF standards
such as SIP, SDP, RTP/RTCP, STUN, TURN, and ICE.
Supports both IPv4 and IPv6, and the following features.
* Audio codecs: AMR, G.711, G.722, G.726, GSM, L16, MPA, OPUS.
* Video codecs: H.263, H.264, H.265, MPEG4, VP8, VP9.
* Audio drivers: Alsa, JACK, OSS, Portaudio, sndio.
* Video sources: FFmpeg avformat, Video4Linux2, X11 Grabber.
* Video output: SDL2, X11, DirectFB.
* NAT Traversal: STUN, TURN, ICE, NATBD, NAT-PMP, PCP.
* Media encryption: SRTP, DTLS-SRTP.
* DNS Service Discovery: Avahi.
* Telemetry messaging: MQTT.
* Control interfaces: JSON-over-TCP.
|
RPM found in directory: /packages/linux-pbone/ftp5.gwdg.de/pub/opensuse/repositories/network:/telephony/15.5/x86_64 |
