What is RTP time out?
Media-inactivity timeout— This parameter indicates the maximum length of time (in seconds) a call can remain active without any media (RTP or RTCP) traffic within a group. Each time an RTP or RTCP packet occurs within a call, this timeout resets. The default setting is 120 seconds.
What is RTP in Asterisk?
The rtp. conf file controls the Real-time Transport Protocol (RTP) ports that Asterisk uses to generate and receive RTP traffic. The RTP protocol is used by SIP, H. 323, MGCP, and possibly other protocols to carry media between endpoints. conf file uses the RTP port range of 10,000 through 20,000.
Which Asterisk configuration file specifies the ports to be used for the media in a SIP call?
conf. Configuration of Asterisk Real Time Protocol, RTP, media channels. RTP is used for SIP communication.
What is host in SIP conf?
host is the domain or host name for the SIP server. This SIP server needs a definition in a section of its own in SIP. 1234 is put into the contact header in the SIP Register message. The contact extension is used by remote SIP server when it needs to send a call to Asterisk.
What is SIP in Asterisk?
The SIP Channel Module enables Asterisk to communicate via VoIP with SIP telephones and exchanges. a SIP server: Asterisk can be configured so that SIP clients (phones, software clients) register to the Asterisk server and set up SIP sessions with the server, i.e. calls and answers incoming calls.
What ports need to be open for Asterisk?
Port ranges for Asterisk:
- For SIP protocol, open UDP (NOT TCP) port 5060 (SIP)
- Open ports 10000-20000.
- Open UDP port 4569 (IAX)
What is Dialplan in Asterisk?
The dialplan is essentially a scripting language specific to Asterisk and one of the primary ways of instructing Asterisk on how to behave. It ties everything together, allowing you to route and manipulate calls in a programmatic way.
What is the difference between Pjsip and Chan SIP?
PJSIP is a separate project, not created or maintained by the Asterisk team. It’s used in many projects, including Asterisk. The chan_sip module uses our own SIP stack and is no longer actively maintained. In the case of Asterisk 13 you need to pass an option to configure[1] to do so.
What are Asterisk channels?
A channel is an entity inside Asterisk that acts as a channel of communication between Asterisk and another device. That is, a phone, a PBX, another Asterisk system, or even Asterisk itself (in the case of a local channel). Channel Drivers provide channels in Asterisk.
How do you check if RTP ports are open?
With the Command Prompt open, type:
- Netstat -ab.
- netsh firewall show state.
- netstat -ano | findstr -i SYN_SENT.
What are RTP ports?
RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features. RTP typically runs over User Datagram Protocol (UDP).
How do I run RTP tests in asterisk?
The majority of RTP tests will be done in the Asterisk testsuite. This involves setting up an RTP session with some remote entity and sending and receiving RTP, testing the accuracy of RTP sent and received, and testing RTCP events for expected statistics.
Why won’t asterisk send RTP keepalives?
Defaults to zero, which means Asterisk won’t send any RTP keepalives: rtpkeepalive=45 rtptimeout(peer) Thistakes as its argument an integer, specified in seconds.
What is the default asterisk retry setting?
This setting defaults to 0, which means that Asterisk will retry indefinitely: registerattempts=0 registertimeout Specifieshow often Asterisk should attempt to re-register to other devices:
Will asterisk update the IP address of a peer upon registration?
If setto yesAsterisk will update the IP address, origination port, and registration period of a peer upon registration. Defaults to yes: rtupdate=yes|no sipdebug Specifies whether or not Asterisk should turn on SIP debugging from the time that Asterisk loads the SIP channel driver: sipdebug=yes|no sendrpid