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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.
SIP trunk unable to connect - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/sip-trunk-unable-to-connect/79088
tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes limitonpeers=yes rtpend=20000 context=from-sip-external callerid=Unknown rtpstart=10000 tcpenable=yes callevents=yes bindport=5062 jbenable=no checkmwi=10 maxexpiry=3600 minexpiry=90 srvlookup=yes tlsenable=no allowguest=yes notifyhold=yes rtptimeout=30 canreinvite=yes …
asterisk-config/sip.conf at master - GitHub
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 Wiki
https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service
The behavior of these parameters is the same as for the SIP TOS settings described above. CHAN_PJSIP In chan_pjsip, there are three parameters that control the TOS settings: a tos option for a type=transport that controls the TOS of SIP signaling packets, a tos_audio option for a type=endpoint that controls the TOS of RTP audio packets, and a …
Tagging traffic leaving FreePBX with QoS value - DSCP ef ...
https://community.freepbx.org/t/tagging-traffic-leaving-freepbx-with-qos-value-dscp-ef-46/34373
In the Chan SIP section of SIP SETTINGS, you can add custom entries (at the bottom - in OTHER SIP SETTINGS): cos_sip = 4 cos_audio = 6 cos_video = 5 tos_sip = cs3 tos_audio = ef tos_video = af41. This is what we use…
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 …
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