At 09:51 2/7/2010, Vinícius Fontes wrote: >You want to set it like this on Asterisk: > >tos_sip=cs3 >tos_audio=ef >tos_video=cs4
Why cs4 instead of af41? > >And in Polycom config: > >qos.ip.rtp.dscp="EF" >qos.ip.callControl.dscp="24" Thanks, Vinícius, but this is for Asterisk v1.4, yes? The current production system is v1.2. > > >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_P>HB_-_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/pro>ducts/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 -- _____________________________________________________________________ -- 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
