BCM and CONTIVITY QoS Help

[carlosmcse]

Hi all,

I'm new to Nortel equipment and VoIP, after a few months using the BCM system I was able to learn allot from it, the problem now is that we implemented VoIP trunks between our offices and want to implement QoS for our network (bad voice quality). We are using Nortel equipment for these WAN connections, Contivity and VPN tunnels the internet router is a Cisco router (we have no control of this router --- ISP router) and all devices are connected to the 5450 Nortel switches. My question is where do I configure the QoS for VoIP? on every device on each LAN? The BCM's? or just the Contivity's?. I assume that if you configure the QoS on one device (BCM or Contivity) the QoS settings will be intercepted by routers (as long they are QoS aware) and pass the parameters along to the other QoS device. Every site has a full T1 to the internet and are connected to the main site and other sites (Mesh Network) via VPN tunnels and OSPF as the routing protocol. I never configured QoS before and I would like to learn more about it and most importantly how to configure the contivity's or BCM's for QoS for VoIP. Can anyone help with this one? This is a medical facility and they just want to try QoS for VoIP only to test out the quality, so to them voice is more important than Data (Patient calls being routed to many locations over the WAN and VoIP).

Thanks,

 

[carlosmcse]

Ok this is what i did on the local BCM for QoS.

Ip Telephony ---> System Configuration:
Echo Cancellation = Enabled w/NLP
G.723.1 Data Rate = 6.3kbps
Reserved Media Gateway Codec = G.711
T.38 UDP Redundancy = 0

Ip Telephony ---> IP Trunks --->H.323
<<<Local Gateway IP Interface>>>
Fallback to Circuit-Switched = Enabled-All
Call signalling = Direct
Primary Gatekeeper IP = 0.0.0.0
Backup Gatekeeper(s) = 0.0.0.0
Alias Names = none
Registration TTL (Seconds) = 60
Gateway Protocol = CSE
H245 Tunneling = Disabled
Call Signaling Port = 1720
Ras Port = 0

<<<Media Parameters>>>
1st Preferred Codec = G.711-uLaw
2nd Preferred Codec = G.729
3rd Preferred Codec = G.723
4th Preferred Codec = None
Silence Compression = Enabled
Jitter Buffer - Voice = Auto
T.38 Fax Support = Disabled
G.729 Payload Size (ms) = 30
G.723 Payload Size (ms) = 30
G.711 Payload Size (ms) = 30

Policy Management --->QoS
Description = BCM QoS Provider
Version = 3.5.0
Status = Enabled
Premium Bandwith % = 60
Video Class = Best Effort
Premium DS Code (0x00-0xFF) = 0xB8
Number of Phone Ports = 4

Policy Management ---> QoS ---> Rules
Name = WP
Destination Address = 192.168.180.5 (Remote BCM)
Destination Address Mask = 255.255.255.0
Source Address = 192.168.170.5 (Local BCM)
Source Address Mask = 255.255.255.0
DSCP(-1^63) = 46 Premium
Protocol = UDP
Destination L4 Port(0-65535) = 0-65535
Source L4 Port(0-65535) = 0-65535
Permit = Yes

** Added this filter to the ip filter group table (WP = Group Name)

Policy Management --> QoS --> Devices
Group Name = LAN1
Queue Set id = WAN
Role Combination = IP-LAN1 (Selected)
Capabilities = Both Selected

Policy management -->Qos --> Actions
Action Name = VoIPAction
Packet Drop = False
Update DSCP(-1^63) = 46 Premium

Policy Management --> QoS --> Policies
Name = WP
FIlter = WP
Filter Type = IP
Interface Group = LAN1
Order = 1
Action = VoIPAction

*** For the remote BCM I did exactly the same thing but the ip addresses for QoS rules (destination and source) are reversed.

I have all my routes configured as a two digit route to the other locations and vice versa but did not enable QoS monitor, because if I do i get a busy signal and the QoS status shows as poor even if I set the threshold to 3 receive and transmit (The reason also I get the busy signal is because I have not setup my PSTN failover) so if i did it would make a PSTN call to the remote location but I just want to use VoIP. I connected to the remote BCM using a i2050 softphone and everything is good so I do I know that QoS is working? I checked the Diagnostics tool provided by the i2050 Softphone and it shows that it's using 802.1p (I setup the softphone to auto).

Here's what I get for the Open RX Stream:
Codec G.711 u-law
Frames/Packet 2 Frames
RTP Port 4294952760
RTCP Port 4294952761
RTP DiffServ 0xb8
RTP 802.1p 6
RTCP DiffServ 0xb8
RTCP 802.1p 6

Here's what I get for the Open TX Stream:
Codec G.711 u-law
Frames/Packet 2 Frames
IP Address 192.168.170.5
RTP Port 28000
RTCP Port 28001
RTP DiffServ 0xb8
RTP 802.1p 6
RTCP DiffServ 0xb8
RTCP 802.1p 6


These two networks are connected via VPN using Contivity switches. Do I have to do anything else or the config I just described above should work?

 

[carlosmcse]

Problem found with above config.

When calling using digitl phones (T7208) over a IP Trunk we can talk and hear the other person with no problem. When we use a IP phone we can make the call over to the other location but I can't hear when the the other party picks up and starts talking and the other prty can't hear me. If the other party (remote office) call me my phone rings but I can't hear anything.

What's up with this? Any ideas?

Thanks

 

[3Sixty]

You have a problem with your VPN config. When a call goes from IP phone to IP phone the call is routed through the BCM for call setup and call alerting. But when the call setup is complete the IP phones talk directly to each other and do not route through the BCM any more. I would confirm you can ping from site to site.I would also make sure the following ports are open.

Signaling between the IP telephones and the Business Communications Manager uses Business Communications Manager port 7000. However, voice packets are exchanged using the default RTP ports 28000 through 28255 at the Business Communications Manager, and ports 51000 through 51200 at the IP telephones. If these ports are blocked by the firewall or NAT, you will experience one-way or no-way speech paths.

Marshall

http://www.marshallshouse.co.uk

 

[acewarlock]

I have not setup my PSTN failover" If this is true, than turn off fallback to Circuit switched

<<<Local Gateway IP Interface>>>
Fallback to Circuit-Switched = Enabled-All ((Disable))

 

[carlosmcse]

THe VPN was verified and everything is good all ports are opened from Site to site. The firewall is not blocking anything on the site to site tunnel. I also tried to use a I2050 softphone to call a digital phone and I have the same problem I can call the digital phone at a remote office (Their phone rings) but I can't hear the other person and the other person can't hear me. Same thing happens when calling from the digitl phone to the Softphone on a different site. If i'm not mistaken when you use a digital phone to call a softphone the communications take place on BCM (Local digital phone) to Remote BCM to establish the call and then it's Softphone to BCM communications. Should I do something with the QoS settings on the Contivities as well or the obove QoS setup is enough on the BCM's only?

 

[3Sixty]

Can you confirm the default gateway on the BCM.

Marshall

http://www.marshallshouse.co.uk

 

[carlosmcse]

Confirmed and I can get to it from any site.

 

[nsantin]

On your contivity, have you enable "Tunnel to Tunnel Traffic" forwaring between dialup tunnels? You need to ensure VPN Client A can connect to VPN Client B. This is off by default.

 

[carlosmcse]

Sure did, all data traffic can forward to all sites (tunnels) with no problem and I'm using OSPF for the routing.

 

[3Sixty]

What are the ping times between sites like?

What code are you using on your IP phones?

Marshall

http://www.marshallshouse.co.uk

 

[3Sixty]

Forgot to ask did you say you could ping from BCM to BCM via the maintenance window.

Marshall

http://www.marshallshouse.co.uk

 

[3Sixty]

Sorry I meant to ask what the ping times where from site to site via the maint window.

http://www.marshallshouse.co.uk

 

[carlosmcse]

Will check it again first thing tomorrow but the average was 15ms.