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asterisk-config/sip.conf at master · RangeNetworks ...
https://github.com/RangeNetworks/asterisk-config/blob/master/sip.conf
tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. cos_sip=3 ; Sets 802.1p priority for SIP packets. cos_audio=5 ; …
IP Quality of Service - Asterisk Project - Asterisk ...
https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service
In chan_sip, there are four parameters that control the TOS settings: "tos_sip", "tos_audio", "tos_video" and "tos_text". tos_sip controls what TOS SIP call signaling packets are set to. tos_audio, tos_video and tos_text control what TOS values are used for RTP audio, video, and text packets, respectively.
Setting ToS in Asterisk | Voip Best Quality
https://qosasterisk.blogspot.com/2015/08/setting-tos-in-asterisk.html#!
What I found interesting is that in sip.conf there are fields for setting ToS field (with default values provided): tos_sip=cs3 (SIP signalling messages) tos_audio=ef (RTP audio) tos_video=af41 (RTP video) tos_text=af41 (RTP text) Settings are also avaliable for other protocols: Even those parameters are named ToS they actually set DSCP. It might be …
Configuring Asterisk
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id216341.html
The last two options, disallow and allow (sip.conf), are used to control which audio codecs are accepted from and offered to the telephone. By defining disallow=all first, we’re telling Asterisk to reset any previously defined codec definitions in the [general] section (or the internal defaults); then we explicitly declare which codecs we’ll accept (and the order we prefer).
asterisk/sip.conf.sample at master · asterisk/asterisk ...
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample
Mirror of the official Asterisk (https://www.asterisk.org) Project repository. No pull requests here please. Use Gerrit: - asterisk/sip.conf.sample at master · asterisk/asterisk
Automatically Receive Call to Play Audio Livestream (MOH ...
https://community.asterisk.org/t/automatically-receive-call-to-play-audio-livestream-moh/83739
I am completely new to Asterisk, but so far, I have managed to install it on my Odroid HC2 (odroidxu4) and edit sip.conf to register my SIP provider. I need help implementing this, and I’m not sure where to start. Here is what I’m trying to do: My church is doing a live stream (audio/video) with OBS, which is eventually pushed to an nginx rtmp server that I set up, which …
One-way audio during outbound calls - Asterisk SIP ...
https://community.asterisk.org/t/one-way-audio-during-outbound-calls/82347
Hello All, I have an Asterisk 16 installation which is running behind a TD-LTE modem-router. I have a DID from my ISP which is configured as a SIP trunk (chan_sip). I have set RTP port range to 7000-20000 in Asterisk also have forwarded this port range in my router. Incoming calls from trunk work well. However when dial an outside number, only outgoing …
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