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How to solve Jiiter Buffer problem in receiving audio RTP ...
https://stackoverflow.com/questions/57177867/how-to-solve-jiiter-buffer-problem-in-receiving-audio-rtp-stream-bad-sound-qual
CrossCompiled Pjsip and Installed all req. libs and sample apps; Sender: Another Windows PC in the same Network using FFmpeg to transmit Audio Stream via Multicast; I got to know about streamutil.c(pjsip sample-apps) which does similar things to send and receive both. Now for the sake of easyness, I'm using the same Cross-Compiled binary ...
Adaptive jitter buffer (2.10) - PJSIP
https://www.pjsip.org/pjmedia/docs/html/group__PJMED__JBUF.htm
The jitter buffer only copied a frame to this buffer when the frame type returned by this function is PJMEDIA_JB_NORMAL_FRAME. p_frm_type. Pointer to receive frame type. If jitter buffer is currently empty or bufferring, the frame type will be set to PJMEDIA_JB_ZERO_FRAME, and no frame will be copied.
Voice pjsip quality issue - General Help - FreePBX ...
https://community.freepbx.org/t/voice-pjsip-quality-issue/50107
ok, I know that for pjsip trunks and extensions there is not jitter buffer configurable parameters in freepbx, in my fanvil phones I can’t find anything related to jitter buffer, but I have enabled VQ RTCP-XR to check a good and …
PJSIP Jitter Buffer - Endpoints - FreePBX Community Forums
https://community.freepbx.org/t/pjsip-jitter-buffer/33594
; PJSIP Jitter Buffer - Test x1000 [from-sip-external] exten => 1000,1,Set(JITTERBUFFER(adaptive)=default) This is literally the only thread on the Internet that deals with FreePBX, PJSIP, and jitter buffers, and I feel like this XKCD comic right now.
Guidelines and Considerations — PJSIP Project 2.10 ...
https://docs.pjsip.org/en/latest/consider.html
The performance of the audio device is probably the one with most issues, as some development boards does not have a decent sound device. Typically there is high audio jitter (or burst) and latency. This will affect end to end audio latency and also the performance of the echo canceller.
Calls — PJSIP Project 2.10 documentation
https://docs.pjsip.org/en/latest/pjsua2/call.html
Working with Call’s Audio Media ... PJSIP_DIALOG_CAP_SUPPORTED if the specified capability is explicitly supported, see pjsip_dialog_cap_status for more info. ... Statistic of raw jitter in receiving direction. It is only used when PJMEDIA_RTCP_STAT_HAS_RAW_JITTER is set to non-zero. RtcpSdes peerSdes. Peer SDES.
Audio stutters while streaming audio - [email protected]
https://pjsip.pjsip.narkive.com/swnOXsJT/audio-stutters-while-streaming-audio
Audio stutters while streaming audio. Ramesh D. 2009-04-27 00:55:21 UTC. Permalink. Hi, I am using streamutil application (that is part of PJSIP 1.1 package) to play streaming audio (PCMU). The audio stutters periodically. This problem occurs even after increasing the value of PJMEDIA_SOUND_BUFFER_COUNT to 16. See below for the output.
https://pjsip.pjsip.narkive.com/7JABDIG3/pjsua-audio-problem
I'm trying to use pjsua to make a call between two linux box, I'm able to. hear audio for few seconds, I'm using alsa, I have the same problem - Using pjsua to call a sip provider. Any. pointers would be appreciated. I can supply logs, if required. in my case I'm using pjsip 1.12, Nicola.
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