WebSep 6, 2016 · Teams. Q&A for work. Connect and share knowledge within a single location that is structured and easy to search. Learn more about Teams WebRFC 5761 Multiplexing RTP and RTCP April 2010 payload types other than 72 and 73 are prohibited when multiplexing RTP and RTCP. This is done to support [], which allows the …
webrtc - how change G722 to PCMA codec? - Stack Overflow
WebApr 23, 2024 · RTCP-mux for Plain Transport RTP IN. mediasoup libraries. friksa (Friksa) April 23, 2024, 8:22pm 1. Using FreeSwitch to bring a phone into a media-soup … WebAug 17, 2024 · but here is problem: when i do orginate user/8801 &echo in fs i got NORMAL_TEMPORARY_FAILURE and no sip message is sent out . seems like fs try to send a websocket request to OPENSIPS_IP:5060 cause transport=ws in contact.. Then i tried to remove transport=ws in contact before opensips send out to fs, this time when i … cantoni wohlen
Using FreeSwitch SIP server to build a video conferencing
WebPost by Miguel Oyarzo ext-rtp-ip ext-sip-ip local-network-acl Those need to be set properly to determine the correct IP to fib about in the SDP, The ACL dictates whats inside the nat all else is outside. Weba=rtcp-mux a=rtpmap:100 VP8/90000 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=rtpmap:96 rtx/90000 a=fmtp:96 apt=100-----send 754 bytes to wss/[208.84.81.64]:57975 at 19:31:56.205982: WebAbout This Book Forget the hassle - make FreeSWITCH work for you Discover how FreeSWITCH integrates with a range of tools and APIs From high availability to IVR development use this book to become more confident with this useful communication software Who This Book Is For If you are a systems admin, a VoIP engineer, a web … canton jewelers