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Media Quality — PJSIP Project 2.10 documentation
https://docs.pjsip.org/en/latest/pjsua2/media_quality.html
Audio Quality¶. If you experience any problem with the audio quality, you may want to try the steps below: Follow the guide: Test the sound device using pjsystest. Identify the sound problem and troubleshoot it using the steps described in: Checking for sound problems. It is probably easier to do the testing using lower level API such as PJSUA since we already have a built-in …
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 a poor quality call …
low sound quality in windows desktop - [email protected]
https://pjsip.pjsip.narkive.com/9K5IK4a8/low-sound-quality-in-windows-desktop
Post by Ebubekir DEMÄ°R. 23:36:44.070 Master/sound !Underflow, buf_cnt=0, will generate 1 frame. The pjsua command cc 0 0 routes the mic audio back to the speaker, hence. there is no codec involved. If this sounds bad then it must be an issue. with the audio device or choice of portaudio/mme/wasapi drivers.
PJSUA-API Media Manipulation (2.10) - PJSIP - Open Source ...
https://www.pjsip.org/pjsip/docs/html/group__PJSUA__LIB__MEDIA.htm
Opaque type of extra sound device, an additional sound device beside the primary sound device (the one instantiated via pjsua_set_snd_dev() or pjsua_set_snd_dev2()).This sound device is also registered to conference bridge so it can be used as a normal conference bridge port, e.g: connect it to/from other ports, adjust/check audio level, etc.
Adaptive Delay Buffer (2.10) - PJSIP
https://www.pjsip.org/pjmedia/docs/html/group__PJMED__DELAYBUF.htm
Once the optimal delay has been learned, the delay buffer will apply this delay to the audio flow, expanding or shrinking the audio samples as necessary when the actual audio samples in the buffer are too low or too high. It does this without distorting the audio quality of the audio, by using PJMED_WSOLA.
Samples: Using SIP and Custom RTP/RTCP to Monitor Quality ...
https://www.pjsip.org/pjmedia/docs/html/page_pjmedia_samples_siprtp_c.htm
This source is an example to demonstrate using SIP and RTP/RTCP framework to measure the network quality/impairment from the SIP call. This program can be used to make calls or to receive calls from other SIP endpoint (or other siprtp program), and to display the media quality statistics at the end of the call.
PJSIP - Open Source SIP, Media, and NAT Traversal Library
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, …
sip - How to get the audio stream from PJSIP when there …
https://stackoverflow.com/questions/46243029/how-to-get-the-audio-stream-from-pjsip-when-there-is-no-audio-hardware-device
I want to use PJSIP's C API to record the incoming audio to a file on a machine with no hardware sound device . I'm unsure about the details, but the sparse documentation for PJSIP suggests it should be possible through the pjsua_set_null_snd_dev() call.
Baku Baku Sans (Better Audio Quality) - YouTube
https://www.youtube.com/watch?v=B4hi2aSLen8
I reuploaded this video cause the old one had bad audio quality. This one does not---So, i probably should've reuploaded this like a year ago, but I kept for...
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