[Asterisk-Users] SIP audio port usage

Sebastian Milioto smilioto at gmail.com
Tue Sep 20 06:50:07 MST 2005


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 at 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 at lists.digium.com
> ->[mailto:asterisk-users-bounces at 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 at 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.
> ->>
> ->>
> ->>
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> ->
> ->
> ->--
> ->Adrien Laurent
> ->adrien at modulis.ca
> ->www.modulis.ca
> ->_______________________________________________
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> ->
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