QoS problem

1. Brekeke Product Name and version:

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:
I have installed the Brekeke PBX version 2.x on a Windows XP.
I have at the moment set on the Proxy side the rule N°1 as follow:

[Matching]
request=^INVITE
TO=sip:(.+)@

[Deploy]


in order to use only the SIP Server side.

I know that the SIP Server can handle packets marked by QoS, while the PBX no, tell me if I'm wrong please.

So I've installed on the same Lan an Audiocodes SIP Gateway Mediant1000 which receives calls from PSTN and route them to the SIP Server side of the Brekeke PBX and then to some IP Phones on the Lan.
The Merdiant1000 is configured to set QoS as follow:
DiffServ 26 for signalling and 46 for rtp flow.

So I have made a call from PSTN and I traced it with Wireshark and on the trace I see the INVITE generated by the Mediant1000with the QoS set, then I see the INVITE packet generated by the SIP Server to the ip phone 106 with no QoS. Why ?
The the rtp relay is set to off because we are in Lan.

The problem is that the SIP Server receives an INVITE packet with the QoS, and re-route this INVITE to the ip phone 106 without QoS.

Can you explain me better how the SIP Server manages the QoS ?

Thank you very much in advance!

Does the sip control packet need to have qos? The rtp stream should have qos but there should be no need for the control packet to have qos. It doesn't matter if the control packet has a 10ms latency or a 300 ms latency.

Yes I think that have the RTP packests with QoS it is enough, but into the INVITE message's session descriprion is stated the QoS.

My problem is to understand in what way the SIP Server or the PBX works regarding the QoS.
I mean:

SIP Server --> does it manages the QoS for RTP and for Signalling?
Brekeke PBX --> does it manages the QoS for RTP and for Signalling?


I need to understand this in order to choice the better way to do things.
Can anyone answers these two questions ,please?

Thank you very much!

What I mean is that I have to set the best QoS Service for this customer who has the following scenario:

- Brekeke PBX at the moment working only with the SIP Server
- Audiocodes Mediant1000 E1 with QoS set on it
- 10 SJPhones on the LAN on which I can't set QoS
- Switch HP on wich I cen set QoS on its ports.

So all the traffic is made on the internal Lan.

Now, what do you suggest to do with your Product in order to have the maximum QoS we can reach in this situation ?

Thank you very much!

Can someone answer me please?

Thank you very much

Flillo,

QOS is typcially used to prioritize traffic going out the WAN. In your case it looks like all the traffice is going to the Audiocodes and then out the E1. Therefore, you should be concerned about prioritizing traffic to the Audiocodes. If all the voip devices/softphones are connected to the switch and the switch has qos, then prioritize udp/rtp traffic on the switch and that should do it.

QoS as defined as Quality of Service; tag line is right to prioritized traffic between hardware devices. It does not matter if this is inside your LAN or WAN. Brekeke does not support QoS as far as I know. voipwell is right your Switch should be good enough to handle the QoS especially if you are routing less than 100 simultaneous calls. Again, it depends what kind of switch you are using, layer3 switch should handle this task better.

If anyone know, if Brekeke support native QoS please post the answer.

>> If anyone know, if Brekeke support native QoS please post the answer.

Go to Brekeke PBX's [Options] => [Advanced] page.
Set "net.rtp.tos" there. It is RTP packet's TOS value.

For example,,
--------------------
net.rtp.tos　= 0x10
--------------------