WebMar 5, 2015 · Create an RTP transport Create a UDP RTP transport structure. Add this structure to the RTP stream structure. Write an init () method and call it from the RTP engine's new () callback. The init () method should create a UDP socket and set any socket level flags required (e.g. nonblocking I/O). 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 …
Asterisk AGI script falls when caller hangup - Stack Overflow
WebJul 8, 2024 · I'm having the simple AGI script, I need to dial 101 extension by calling 6666 number and calculate answered time after call. Everything works fine when callee hangup, but when caller hangup agi script falls with returning 4. http://forums5.grandstream.com/t/explanations-on-strict-rtp-parameter-in-rtp-settings/39555 mrsa cdcガイドライン
Explanations on Strict RTP parameter in RTP Settings
Web> 0x7f01780564d0 -- Strict RTP learning after ICE completion > 0x7f0178020420 -- Strict RTP learning after ICE completion > 0x7f0178020420 -- Strict RTP switching to RTP target address 10.176.17.207:54799 as source > 0x7f01780564d0 -- Strict RTP switching to RTP target address 10.176.17.207:54789 as source WebOct 22, 2024 · The RTP might not be arriving at Asterisk, as tcpdump captures before the the Linux firewall, so please use “rtp set debug on” at the Asterisk CLI, to see if the RTP is actually arriving (and if it is being forwarded. Is there a valid route to 10.20.P.A? I notice you are offering G.729. 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): mrsa とは 感染経路