Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca
It depends on the ATA, and our router, etc... Typically in the range between 10000 and 20000 ->-----Original Message----- ->From: asterisk-users-bounces@lists.digium.com ->[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ->Adrien Laurent ->Sent: Monday, September 19, 2005 12:23 PM ->To: asterisk-users@lists.digium.com ->Subject: [Asterisk-Users] SIP audio port usage -> ->Hi, -> ->I know that SIP is using port 5060 for session initiation, ->but which port does it use for audio ? is it dynamically assigned ? -> ->Thanks, -> ->Adrien -> ->-- ->Adrien Laurent - CIO ->www.modulis.ca ->514-284-2020 ext 202 ->adrien@modulis.ca -> -> ->_______________________________________________ ->--Bandwidth and Colocation sponsored by Easynews.com -- -> ->Asterisk-Users mailing list ->Asterisk-Users@lists.digium.com ->http://lists.digium.com/mailman/listinfo/asterisk-users ->To UNSUBSCRIBE or update options visit: -> http://lists.digium.com/mailman/listinfo/asterisk-users ->
On Mon, Sep 19, 2005 at 12:22:35PM -0400, Adrien Laurent wrote:> I know that SIP is using port 5060 for session initiation, but which port > does it use for audio ? is it dynamically assigned ?RTP protocol is used for audio. Port range is defined in /etc/asterisk/rtp.conf -- Stefan Tichy <asterisk@pi4tel.de>
> I know that SIP is using port 5060 for session initiation, but which port > does it use for audio ? is it dynamically assigned ?Its dynamically assigned on a per-call basis. Asterisk assigns the port based on contents of rtp.conf. Remote sip phones assign port numbers based on whatever the manufacturer happened to choose (no industry standard). E.g., Cisco uses 32,768 to something around 40,000, while xlite uses something in the area of 8,000. The various manufacturers are not consistent at all.
So the more reliable way to do QoS is with MAC adress and not on a port basis. Am I right ? Thanks for your help, Adrien On 9/19/05, Rich Adamson <radamson@routers.com> wrote:> > > I know that SIP is using port 5060 for session initiation, but which port > > does it use for audio ? is it dynamically assigned ? > > Its dynamically assigned on a per-call basis. > > Asterisk assigns the port based on contents of rtp.conf. > > Remote sip phones assign port numbers based on whatever the manufacturer > happened to choose (no industry standard). E.g., Cisco uses 32,768 to > something around 40,000, while xlite uses something in the area of 8,000. > The various manufacturers are not consistent at all. > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Adrien Laurent adrien@modulis.ca www.modulis.ca
Yes, because then the MACs specified would be getting the QoS, not just certain ports. This is how I set up my customers when they have QoS available. ->-----Original Message----- ->From: asterisk-users-bounces@lists.digium.com ->[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ->Adrien Laurent ->Sent: Tuesday, September 20, 2005 8:53 AM ->To: Asterisk Users Mailing List - Non-Commercial Discussion ->Subject: Re: [Asterisk-Users] SIP audio port usage -> ->So the more reliable way to do QoS is with MAC adress and not ->on a port basis. ->Am I right ? -> ->Thanks for your help, -> ->Adrien -> ->On 9/19/05, Rich Adamson <radamson@routers.com> wrote: ->> ->> > I know that SIP is using port 5060 for session ->initiation, but which ->> > port does it use for audio ? is it dynamically assigned ? ->> ->> Its dynamically assigned on a per-call basis. ->> ->> Asterisk assigns the port based on contents of rtp.conf. ->> ->> Remote sip phones assign port numbers based on whatever the ->> manufacturer happened to choose (no industry standard). E.g., Cisco ->> uses 32,768 to something around 40,000, while xlite uses ->something in the area of 8,000. ->> The various manufacturers are not consistent at all. ->> ->> ->> ->> _______________________________________________ ->> --Bandwidth and Colocation sponsored by Easynews.com -- ->> ->> Asterisk-Users mailing list ->> Asterisk-Users@lists.digium.com ->> http://lists.digium.com/mailman/listinfo/asterisk-users ->> To UNSUBSCRIBE or update options visit: ->> http://lists.digium.com/mailman/listinfo/asterisk-users ->> -> -> ->-- ->Adrien Laurent ->adrien@modulis.ca ->www.modulis.ca ->_______________________________________________ ->--Bandwidth and Colocation sponsored by Easynews.com -- -> ->Asterisk-Users mailing list ->Asterisk-Users@lists.digium.com ->http://lists.digium.com/mailman/listinfo/asterisk-users ->To UNSUBSCRIBE or update options visit: -> http://lists.digium.com/mailman/listinfo/asterisk-users ->
But, if I have Xlite running on client PC and at the same time the user is doing FTP, both service has the same QoS treatment? Is there a way to differentiate these services besides the port? Sebastian On 9/20/05, Sherwood McGowan <madprofzero@yahoo.com> wrote:> Yes, because then the MACs specified would be getting the QoS, not just > certain ports. This is how I set up my customers when they have QoS > available. > > ->-----Original Message----- > ->From: asterisk-users-bounces@lists.digium.com > ->[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > ->Adrien Laurent > ->Sent: Tuesday, September 20, 2005 8:53 AM > ->To: Asterisk Users Mailing List - Non-Commercial Discussion > ->Subject: Re: [Asterisk-Users] SIP audio port usage > -> > ->So the more reliable way to do QoS is with MAC adress and not > ->on a port basis. > ->Am I right ? > -> > ->Thanks for your help, > -> > ->Adrien > -> > ->On 9/19/05, Rich Adamson <radamson@routers.com> wrote: > ->> > ->> > I know that SIP is using port 5060 for session > ->initiation, but which > ->> > port does it use for audio ? is it dynamically assigned ? > ->> > ->> Its dynamically assigned on a per-call basis. > ->> > ->> Asterisk assigns the port based on contents of rtp.conf. > ->> > ->> Remote sip phones assign port numbers based on whatever the > ->> manufacturer happened to choose (no industry standard). E.g., Cisco > ->> uses 32,768 to something around 40,000, while xlite uses > ->something in the area of 8,000. > ->> The various manufacturers are not consistent at all. > ->> > ->> > ->> > ->> _______________________________________________ > ->> --Bandwidth and Colocation sponsored by Easynews.com -- > ->> > ->> Asterisk-Users mailing list > ->> Asterisk-Users@lists.digium.com > ->> http://lists.digium.com/mailman/listinfo/asterisk-users > ->> To UNSUBSCRIBE or update options visit: > ->> http://lists.digium.com/mailman/listinfo/asterisk-users > ->> > -> > -> > ->-- > ->Adrien Laurent > ->adrien@modulis.ca > ->www.modulis.ca > ->_______________________________________________ > ->--Bandwidth and Colocation sponsored by Easynews.com -- > -> > ->Asterisk-Users mailing list > ->Asterisk-Users@lists.digium.com > ->http://lists.digium.com/mailman/listinfo/asterisk-users > ->To UNSUBSCRIBE or update options visit: > -> http://lists.digium.com/mailman/listinfo/asterisk-users > -> > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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