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IP Quality of Service - Asterisk Project - Asterisk ...
https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service
The TOS byte is used by the network to provide some level of Quality of Service (QoS) even if the network is congested with other traffic. Asterisk running on Linux can also set 802.1p CoS marks in VLAN packets for the VoIP protocols it uses. This is useful when working in a switched environment. In fact Asterisk only set priority for Linux socket.
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 …
No audio when endpoint and Asterisk are both behind ...
https://community.asterisk.org/t/no-audio-when-endpoint-and-asterisk-are-both-behind-distinct-nats/75871
Below are relevant segments from various config files, I hope I covered the important areas. **pjsip.endpoint.conf** [301] type=endpoint aors=301 auth=301-auth tos_audio=ef tos_video=af41 cos_audio=5 cos_video=4 allow=ulaw,alaw,gsm,g726,g722 context=from-internal callerid=My Mobile Phone Extension <301> dtmf_mode=rfc4733 …
Asterisk problème d'appel entrant - Asterisk SIP ...
https://community.asterisk.org/t/asterisk-probleme-dappel-entrant/91090
Bonjour je suis entrain de mettre en place une configuration asterisk mais j’arrive pas a faire fonctionner les appelles entrant. je vous met ci joint mon sip.conf: [general] nat=no context=atmos-default-context defaultexpiry=600 udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 transport=udp srvlookup=no tos_sip=cs3 tos_audio=ef …
TOS QoS Values Not Changing - FreePBX Community Forums
https://community.freepbx.org/t/tos-qos-values-not-changing/42765
The following values are present in sip_general_additional.conf in the latest stable FreePBX 14 Asterisk 13 release. tos_sip=cs3 tos_audio=ef tos_video=af41. Adding new values to Settings–>SIP Settings–>Chan-SIP does not seem to override these values. Also, a wireshark analysis of the UDP packets reveals they are being sent with a DSCP tag of 0x05.
How to disable QoS in Asterisk 11.9? - Asterisk - Helperbyte
https://helperbyte.com/questions/299965/how-to-disable-qos-in-asterisk-119
Good afternoon. Inters the issue forced off QOS in Asterisk to 11.9. tos_sip=cs3 tos_audio=ef cos_sip=3 cos_audio=5 The log gives the following:... == Using SIP TOS bits 96
asterisk-config/sip.conf at master · RangeNetworks ...
https://github.com/RangeNetworks/asterisk-config/blob/master/sip.conf
udpbindaddr=0.0.0.0 ; asterisk 1.8; IP address to bind UDP listen socket to (0.0.0.0 binds to all); Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio …
QoS / ToS / DSCP not configurable and defaulting to no ...
https://community.freepbx.org/t/qos-tos-dscp-not-configurable-and-defaulting-to-no-priority-with-pjsip/34902
L.S., I found that the latest distro/FreePBX core release does not set PJSIP transport tos or endpoint tos_audio/tos_video and as far as I can see doesn’t allow for a webgui method of setting these parameters. As result all SIP and RTP traffic from the PBX is not and can not be prioritized on the network when using PJSIP, which is bad practise. A temporary …
Commlite Auto-Focus Adapter Canon EF to Sony NEX Mount ...
https://www.nardrental.com/commlite-auto-focus-adapter-canon-ef-to-sony-nex-mount.html
Commlite new-generation electronic lens mount adapter CM-EF-NEX is mainly used for connecting the ‘Canon EF series lens’ to ‘Sony NEX series cameras with E-mount’. Meanwhile, Commlite utilizes the new electronic technology, which can make the aperture exactly being aligned via the adapter, and adjusted for auto-focus.
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