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: [EMAIL PROTECTED] 
->[mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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.
->>
->>
->>
->> _______________________________________________
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->
->
->--
->Adrien Laurent
->[EMAIL PROTECTED]
->www.modulis.ca
->_______________________________________________
->--Bandwidth and Colocation sponsored by Easynews.com --
->
->Asterisk-Users mailing list
->[email protected]
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->


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