We have collected the most relevant information on Asterisk Tos_Audio. Open the URLs, which are collected below, and you will find all the info you are interested in.


SIP - Asterisk: The Definitive Guide (3rd edition)

    http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-A-SECT-2.html
    tos_sip, tos_audio, and tos_video Asterisk can set the TOS bits in the IP header to help improve performance on routers that respect TOS bits in their routing calculations. The tos_sip , tos_audio , and tos_video settings control the TOS bits for the …

Pjsip : DSCP tos & cos - Asterisk SIP - Asterisk Community

    https://community.asterisk.org/t/pjsip-dscp-tos-cos/76267
    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 tos_video option for a type=endpoint that controls the TOS of video packets.

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.

Asterisk Audio and Video Capabilities - Asterisk Project ...

    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Audio+and+Video+Capabilities
    Asterisk supports a variety of audio and video media. Asterisk provides CODEC modules to facilitate encoding and decoding of audio streams. Additionally file format modules are provided to handle writing to and reading from the file-system. The tables on this page describe what capabilities Asterisk supports and specific details for each format.

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
    Hello, I’m relatively new to Asterisk so please go easy 🙂 The problem In certain conditions, audio is not working when a call is established between two endpoints. For debugging purposes I’ve narrowed down the scenario to a call from a soft client to Asterisk (*60—talking clock). The Setup Both Asterisk and a soft client are behind two distinct NATs. Asterisk is …

SIP RTP TOS bits 184 in TCLASS field - Asterisk Support ...

    https://community.asterisk.org/t/sip-rtp-tos-bits-184-in-tclass-field/41342
    Hello all, I have a production server which I just upgrade from v.1.8 to 11, where the ToS I have it configured as: [quote]tos_sip=cs3 tos_audio=ef cos_sip=3 cos_audio=5[/quote] But in the console I see that the using of the SIP ToS is failing, as: [quote] == Using SIP RTP TOS bits 184 == Using SIP RTP TOS bits 184 in TCLASS field. == Using SIP RTP CoS mark 3[/quote] I’m trying to figure …

Curso Asterisk (VI): Lidiando con el NAT | The Infamous ...

    https://www.axelko.com/techblog/2014/03/curso-asterisk-vi-lidiando-con-el-nat/
    Curso Asterisk (VI): Lidiando con el NAT. El enemigo público número uno del protocolo SIP son las tablas NAT. El NAT es la principal causa de problemas a la hora de montar nuestro servidor Asterisk. Desafortunadamente para nosotros, debido a la falta de IPs públicas de IPv4, lo normal en nuestros hogares es que estemos detrás de un NAT.

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. I cannot …

QoS sobre nuestras comunicaciones VoIP (SIP y RTP) - Asterisk

    https://netvoip.wordpress.com/2015/04/29/qos-sobre-nuestras-comunicaciones-voip-sip-y-rtp/
    QoS sobre nuestras comunicaciones VoIP (SIP y RTP) Cuando etiquetamos tráfico con diferentes priorirdades, existen dos modos de hacerlo, a nivel de protocolo IP (ToS) o a nivel de VLAN (802.1p). En este post vamos a ocuparnos de la primera parte, de la que se encarga de realizar diferenciación a nivel de IP (ToS).

asterisk:realtime-integration - Kamailio (OpenSER) Wiki

    https://kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
    This is a tutorial on how to integrate OpenSER with Asterisk v1.2 and the new realtime functions. First, create the views. This allows you to use the same users you already had without having to manually replicate them into another database.

Now you know Asterisk Tos_Audio

Now that you know Asterisk Tos_Audio, we suggest that you familiarize yourself with information on similar questions.