
RTP Blog Articles
Asterisk MulticastRTP channels
MulticastRTP Asterisk 1.8: A new RTP engine and channel driver have been added which supports Multicast RTP. The channel driver can be used with the Page application to perform multicast RTP paging. The dial string format is: MulticastRTP/<type>/<destination>/<control…
Read More »ZRTP
ZRTP is key exchange protocol designed to enable VoIP devices to agree keys for encrypting media streams (voice or video) using SRTP. ZRTP is defined in an Internet draft https://datatracker.ietf.org/doc/html/draft-zimmermann-avt-zrtp. The authors of ZRTP describe it as “Media…
Read More »Kamailio 1.5.x and RTPProxy
This is a basic configuration file for Kamailio (OpenSER) v1.5.x to do VoIP NAT traversal using RTPProxy. valid for Kamailio (OpenSER) 1.5.x # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with:…
Read More »Asterisk sip directrtpsetup
Version Introduced in Asterisk 1.4 Description directrtpsetup=yes is similar to directmedia=, except the audio is redirected in the initial INVITEs rather than reinviting the media a few RTP packets in. Note: canreinvite= was renamed to directmedia= in Asterisk…
Read More »About RTPproxy
The RTPproxy is a high-performance software proxy for RTP streams that can work together with SIP Express Router or OpenSER. Originally created for handling NAT scenarious it can also act as a generic media relay as well as…
Read More »Asterisk SRTP
General info about SRTP can be found here Asterisk 1.8 has native support for SRTP ! Ref. http://bugs.digium.com/view.php?id=5413 Review board http://reviewboard.digium.com/r/191/, http://lists.digium.com/pipermail/asterisk-dev/2009-January/036029.html You MUST secure signaling because SRTP keys are exchanged in plaintext with SDES. Use TLS –…
Read More »RTP Ports
Realtime Transport Protocol. RTP in general is described in RFC 3550. This range is not registered (it never could be, being so broad) but it seems to be somewhat common. See Are there specific ports assigned to RTP?…
Read More »Converting RTP to audio
For debugging and troubleshooting it is often desirable to convert a stream of captured RTP packets to playable audio (e.g. a WAV file). A major difficulty in doing this is decoding the many different types of Codecs that…
Read More »SRTP
SRTP is a security profile for RTP that adds confidentiality, message authentication, and replay protection to that protocol. It is an action item in the IETF Audio-Video Transport Working Group, where it is an Internet Draft and is…
Read More »RTP
RTP opens two ports for communication. One for the media stream (an even port number) and one for control (QoS feedback and media control) – RTCP. The port numbers are not hard defined, it depends very much upon…
Read More »Asterisk sip rtptimeout
rtptimeout Terminate call if 60 seconds of no RTP activity when we’re not on hold. Added in June 2004 to CVS-HEAD Example rtptimeout=60 See also >Asterisk sip rtpholdtimeout AbsoluteTimeout Note from MarkSter’s writing on bugtrack: However, I’ve added…
Read More »RTP Silence Suppression
Endpoints sending audio as an RTP stream are not required to send packets during silent periods. The capability to stop sending RTP packets during silent periods is known as “Silence Suppression” or VAD (Voice Activity Detection). Whether to…
Read More »RTP Symmetric
Symmetric RTP means that the UA uses the same socket/port for sending and receiving the RTP stream. How to set it up is defined in the SDP part of a SIP Invite. An IETF Internet draft for connection-oriented…
Read More »Asterisk config rtp.conf
rtp.conf Configuration of Asterisk Real Time Protocol, RTP, media channels. RTP is used for SIP communication. Details On your router you might want to arrange both traffic shaping (QoS) and port forwarding (in case of NAT) for the…
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