[asterisk-users] TOS bits, DSCP, Asterisk & Polycom

Vinícius Fontes vinicius at canall.com.br
Sun Feb 7 09:51:40 CST 2010


You want to set it like this on Asterisk:

tos_sip=cs3
tos_audio=ef
tos_video=cs4

And in Polycom config:

qos.ip.rtp.dscp="EF"
qos.ip.callControl.dscp="24"


Atenciosamente,

Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000

Information Security Manager
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brazil
+55 54 2104-7000

----- "Doug" <Doug at NaTel.net> escreveu:

> Has anyone figured this out yet?
> 
> Lots of places say to add the following
> to sip.conf of an Asterisk 1.2 system
> (current production machine/Asterisk as root):
> 
>    tos=0xB8
> 
>    (Hex B8 = Decimal 184 = Binary 10111000)
> 
> or if you are running Asterisk v1.4 or newer:
> 
>    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.
> 
> 
> To match the current 1.2 machine would I set the Polycom's
> sip.cfg to the first or second QOS option?
> 
> Option 1:
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>     <QOS>
>        <Ethernet>
>           <RTP
>             qos.ethernet.rtp.user_priority="5"/>
>           <CallControl
>             qos.ethernet.callControl.user_priority="5"/>
>           <Other qos.ethernet.other.user_priority="2"/>
>        </Ethernet>
> 
>        <IP>
>           <RTP
>             qos.ip.rtp.dscp=""
>             qos.ip.rtp.min_delay="1"
>             qos.ip.rtp.max_throughput="1"
>             qos.ip.rtp.max_reliability="1"
>             qos.ip.rtp.min_cost="0"
>             qos.ip.rtp.precedence="5"/>
> 
>           <CallControl
>             qos.ip.callControl.dscp=""
>             qos.ip.callControl.min_delay="1"
>             qos.ip.callControl.max_throughput="1"
>             qos.ip.callControl.max_reliability="1"
>             qos.ip.callControl.min_cost="0"
>             qos.ip.callControl.precedence="5"/>
>        </IP>
>     </QOS>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> 
> 
> Option 2:
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>     <QOS>
>        <Ethernet>
>           <RTP
>             qos.ethernet.rtp.user_priority="5"/>
>           <CallControl
>             qos.ethernet.callControl.user_priority="5"/>
>           <Other qos.ethernet.other.user_priority="2"/>
>        </Ethernet>
> 
>        <IP>
>           <RTP
>             qos.ip.rtp.dscp="ef"
>             qos.ip.rtp.min_delay="1"
>             qos.ip.rtp.max_throughput="1"
>             qos.ip.rtp.max_reliability="1"
>             qos.ip.rtp.min_cost="0"
>             qos.ip.rtp.precedence="5"/>
> 
>           <CallControl
>             qos.ip.callControl.dscp="ef"
>             qos.ip.callControl.min_delay="1"
>             qos.ip.callControl.max_throughput="1"
>             qos.ip.callControl.max_reliability="1"
>             qos.ip.callControl.min_cost="0"
>             qos.ip.callControl.precedence="5"/>
>        </IP>
>     </QOS>
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> 
> or none of the above?
> 
> Also, how does "10111000" Fit into:
> 
>                [ 0   1   2  ]  [3]  [4]  [5]  [6       7]
>                [ Precedence ]  [D]  [T]  [R]  [ECN Field]
> 
> Is it read backwards?
> 
> Any helpful comments appreciated.
> 
> References:
> 
>    <http://en.wikipedia.org/wiki/Type_of_Service#Type_of_Service>
> 
>   
> <http://en.wikipedia.org/wiki/DiffServ#Expedited_Forwarding_.28EF.29_PHB_-_DSCP.3D.2846_OR_101110.29>
> 
>    <http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+tos>
> 
>   
> <http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/SoundPointIP_SoundStationIP_AdminGuide_SIP3_0_Eng_Rev_A.pdf>
> 
> 
> 
> 
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list