[Asterisk-Users] SIP audio port usage

[Adrien Laurent]

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