Stephane M wrote:
>
> I am bit surprised that the QoS is not so good actually...

The people who make routers seem to understand very very little about SIP.

If you really want Qos to work, stop faffing and buy a CTX1000.

> I thought that SIP was the protocol for all VoIP communications ? I
> mean, I thought that even the the Audio of the communciation, used this
> protocol... !?!

Not at all.

Session initiation protocol. SIP sets up the sessions. In the most
common case, they are 2 RTP audio sessions (one each way). The sessions
are described using SDP - which is carried in the SIP messages.

In practice SIP could setup anything. The ones I've heard talked about
are online gaming sessions and shared whiteboard sessions.


> Anyway, I found on ny phone, that RTP port is [ 16384 - 16482 ]
> So I added, on my QoS service..

Those will be the ports for which your phone requests RTP to be sent to.
Your inbound RTP.

Your SIP service provider may request that the RTP is sent outbound to
completely different ports.

> 1) Do you know which protocol is using RTP ? (TCP/IP ???)

RTP travels over UDP which travels over IP.


> Concerning a previous 'post' on Secure VoIP, I didn't find anything on
> SIPS, but found into the "line EXT" the SRTP private Key
> Now, I will investigate on an ASterisk server, how I can add the key for
> the communication...

For SRTP, you really need to use a new random key for each call.

I can run SIP training days if you want to learn a lot more quickly.


Tim