WebFeb 16, 2024 · Asterisk WebRTC outgoing call delay. I run an Asterisk 16 installation and a WebPhone based on SIP.js. Unfortunately, I often don't hear the first few seconds when I call someone. But everything is fine with incoming calls. The Asterisk is in a data center, the browser / client is behind NAT. [Nov 2 17:58:11] VERBOSE [15217] [C-00000002] app ... WebAug 29, 2024 · They work in conjunction with one another, so it’s important to understand their unique roles. RTCP is Real Time Control Protocol, which works alongside RTP to …
View topic - Asterisk keep crashing frequently in a day - eflo
Web[11:29:56] > 0x7f91c8032e50 -- Strict RTP learning after remote address set to: 000.00.136.133:37113 [11:29:57] == Manager 'sendcron' logged off from 127.0.0.1 … peter\u0027s brother in the bible
sip - Asterisk WebRTC outgoing call delay - Server Fault
WebOct 22, 2024 · Below is the most important part (once extension 2204 answered the transferred call) of the sip debug and below it the detailed log starting from sending the call for the agent until the end of transfer for the extension 2204: The summary sip debug (once extension 2204 answered the call and until the left 'simple_bridge' basic-bridge debug): WebDec 22, 2024 · The “strictrtp” option in rtp.conf enables a feature of the RTP stack that learns the source address of media for a session and drops any packets that do not … WebAug 29, 2024 · I'm trying to run Asterisk 16 behind NAT, but it happens that when I make the connection, I can't get any audio. Initially, I realized that Asterisk was receiving from JsSIP the IP of the local machine, I suspected this would be the problem and I deployed the Google stun, the IP problem was fixed but the audio remained mute. peter\u0027s bump clinic nh