Configure BCM 400 VOIP to CS 1000

[MagnuM2K]

Hi,

I'm currently trying to get better VOIP performance from my BCM 400 <--> CS 1000 configuration.

I wasn't the person that originally configured it, but I've checked out all the changes on my BCM 400 and also the CS 1000 and it seems to be configured for SIP routing, but I also see elements of H.323.

My ultimate goal is to get better VOIP performance and see whether it is possible to route VOIP calls over H.323. I've read that H.323 uses TCP, while SIP uses UDP. Would it make a difference in routing and packet lose?

What should I be looking at first? What information should I post here so that I can get an idea whether the settings currently in place are correct.

Thanks

 

[Dasheen]

What problems r u experiencing?

 

[Deweyhumbolt]

Considering it's Nortel to Nortel, I'd prefer H323.

Adversity is Opportunity

 

[MagnuM2K]

At times the VOIP call is very fuzzy/staticy, while at other times it's fine. We only have 4 VOIP trunks, but I would like to ensure that quality is good.

How you I go about configuring it for H.323 only? Or make sure that traffic is routed over H.323 first? Am I right in thinking that quality would be better of H.323?

Thanks for your help.

 

[MagnuM2K]

In addition, my Keycodes show that I have 4 SIP GW Trunks.. Do I need another or different keycode if I want to configure H.323?

 

[Deweyhumbolt]

Yes, those are SIP only.

Adversity is Opportunity

 

[Dasheen]

Quality is affected by many things like bandwidth. What codec r u using? G711 is the best. Are both systems using the same payload? Default on the BCM is 20mS and CS1k is 30mS. Both systems should match. Do you have Vlans setup on the network...one for voice and one for data?

 

[MagnuM2K]

What keycodes would I need if I'd want to get this working over H.323? Is it even worth it? Is there a cost associated with the keycodes?

No VLANs, currently.

These are my settings on the BCM 400.

Under, media gateways, it says G.711 for reserved media gateway codec.

Under, telephony resources, IP Trunks H323 Settings,
Telephony Settings - Enabled-All, Gateway protocol - None.
Under, Configuration I have my gateway IP, Call signaling, Gatekeeper Router, and other settings...

H323 Media Parameters,
G.729
G.723
G.711-uLaw
G.711-aLaw

In that order..

Then Settings,
Jitter - Auto
G.729 payload size - 30
G.723 payload size - 30
G.711 payload size - 30

All other options are off.

Then SIP Settings,
Telephony Settings - Enabled-All, SIP settings, my domain, Call signalling port - 5060, Outgoing Transport - UDP (I see TCP, can this be changed without beaking anything.. Does the CS1000 end also have a TCP setting?

Then SIP Media Parameters,
Preferred Codecs
G.729
G.711-uLaw
G.723
G.711-aLaw

In that order..

Then Settings,
Enable Voice Activity Detection - Checked
Jitter - Auto
G.729 payload size - 20
G.723 payload size - 30
G.711 payload size - 20
Enable T.38 fax - Checked

There are all my settings on my BCM 400 side.. What would I need to check on the CS1K side?

Thanks

 

[Dasheen]

If ur using your network for both data and voice u should segment the network into Vlans.

If you want to use H323 trunking, u'd have to purchase the keycodes.

 

[MagnuM2K]

Am I right in thinking that I'll get better performance if I use H323, or am I going about it all wrong?

 

[subx]

Nope the only difference will be that you will get more Features since there is MCDN over H323 possible.

For the performance its more important what codec or payload you will use as well as bandwidth and of course Quality of Service... That would be the first thing you should integrate!

cheers,

subx

 

[MagnuM2K]

Ok. Where should I start looking for the codec, payload and QOS on the CS1K and also the BCM 400. I'm a little more familiar with the BCM, but I would like to match the changes done on the CS1K and the corresponding location on the BCM 400.

Thanks

 

[Dasheen]

As quoted by GWebster in another thread:

There are three network characteristics that affect voice quality:
- delay (how long it takes packets to traverse the network)
- packet loss (how many packets get lost along the way)
- jitter (variation in how long packets traverse the network)

Unfortunately, voice requires packets to arrive at a constant rate with a fixed delay between those packets. If the next packet is not received in time, it has to be dropped and considered as lost. If jitter is high, whole groups of packets can be late or lost and result in garbled speech. All networks have some jitter and built-in "jitter buffers" handle those variations.

 

[Deweyhumbolt]

The first thing to check as Dasheen noted is your payload size. The defaults are different between the BCMs and CS1Ks.
Next, everything under H323 is useless because you don't have it and it's not worth buying those keycodes. Your Codec choice is G729 first and G711 second. If you want the best quality, put G711 first. Finally you need to setup VLANs.

Adversity is Opportunity

 

[subx]

paylod should be the same as on the CS1000 as mention in this thread already... change it on the BCM System since the CS1000 is always good for a Master ;O)

Codecs.. You should use G729 .. G723 isnt really that good quality at all... G711 will probably not work because of your bandwidth...

QoS you need to enable on BCM and as well on CS1000 but the most important is the network itself.... Its needed that the switches and routers are able to provide QoS and are configured for those things...

 

[SupportDude]

I'm surprised that nobody asked this yet ... but, how much bandwidth do you have between the CS1K and the BCM?
Is there any other traffic on this span besides the VoIP?

-SD-

http://www.mysupportguy.net

 

[Dasheen]

SD I kinda suggested that and also raised the issue of segmenting the network into VLANS. Sounds like part of the problem is BW and both voice & data are fighting over the pipe all at once w/ the voice not getting priority.

 

[MagnuM2K]

At my end, I have a 1.5 Mbps max up/down wireless connection, while at the other office it is a T1 connection. Traffic is not separated in VLANs at the moment. My end only has VOIP and occasional Vid conferencing on that cct. While the end, is our primary link to the internet, thus it shares traffic with all data.

I think I can find all those settings on the BCM, but where would I look on the CS1K, would it be in NRS?

Would I put G.711-uLaw or aLaw 1st?

 

[SupportDude]

Sorry Dash! I missed it.

You should probably use G729 to save bandwidth.
Your issue could be latency/delay with the up/down link on your wireless.
You definitely need to prioritize your voice traffic.
The sets will automatically mark the DSCP at 46 for Priority Queuing.
And as Dasheen pointed out, VLAN's are not a bad idea either.

I believe your Sig Server contains all the settings on the CS1K side.

-SD-

http://www.mysupportguy.net

 

[Dasheen]

No problem SD. U didn't hurt my feelings. Easy to miss things when glancing thru all the comments.