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" <[email protected]> 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 -- _____________________________________________________________________ -- 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
