Echo on meet-me conference

[telecomadmin12]

I have a meet-me vdn assigned. People call in from outside on a co trunk, as well as from inside. When I use it I have a lot of echo. What could be the reason for it?

 

[montyzummer]

the outside co would be my first thought or users using a mobile phone stood next to a deskphone whilst in the conf ,

what happens if you use 5 phones in different areas(all internal) add one by one then dial in on the trunk

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[telecomadmin12]

We were using a conference phone. One party called in on an analog CO trunk. Another party called in on a H.323 trunk. When the party calling from the H.323 hung up, the echo went away.
How do I control echo cancellation on a CO and a H.323 trunk?

 

[montyzummer]

What version of Cm?

What type of CO trunk as in cm 3 there was a known issue between ds1 and ip calls

So that looks like some issue with the h.323 (obviously) there are lots of variables here , can you replicate this and take a trace

When you say h.323 trunk is that another system connected to your CM ??

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[telecomadmin12]

CM3.0
I have a few remote extensions on a VoIP PBX (Asterisk), which I connect to Avaya over H.323 trunks.
Below a trace on the H.323 while a call was engaged between a remote client and a meet-me vdn with a lot of echo on the Avaya end (nothing on my remote SIP client). No speakerphone involved.
I have a medpro TN2302AP. Could that be the issue and would that need some echo cancellation?


09:53:05 G711MU ssff ps:20 rn:1/1 10.1.1.184:10662 10.1.1.186:3000
09:53:05 xoip: fax:Relay modemff tty:US 10.1.1.186:3000 uid:0x5002f
09:53:07 active announcement 5890 cid 0x33ea
09:53:07 hear annc board 04B17 ext 5890 cid 0x33ea
09:53:11 idle announcements cid 0x33ea
VOIP data from: 10.1.1.186:3000
09:53:16 Jitter:32 16 26 16 22 19 25 16 15 0: Buff:95 WC:206 Avg:20
09:53:16 active announcement 5896 cid 0x33ea
09:53:16 hear annc board 04B17 ext 5896 cid 0x33ea
09:53:16 Pkloss:0 0 0 0 0 0 0 0 0 *: Oofo:0 WC:0 Avg:0
09:53:20 idle announcements cid 0x33ea
VOIP data from: 10.1.1.186:3000
09:53:26 Jitter:18 39 0 19 18 18 20 18 20 21: Buff:29 WC:255 Avg:21
09:53:26 Pkloss:0 0 * 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0
VOIP data from: 10.1.1.186:3000
09:53:36 Jitter:18 18 20 19 20 18 19 20 18 19: Buff:35 WC:255 Avg:20
09:53:36 Pkloss:0 1 0 0 0 0 0 0 0 0: Oofo:0 WC:1 Avg:0
VOIP data from: 10.1.1.186:3000
09:53:46 Jitter:19 21 19 22 20 21 21 20 18 19: Buff:32 WC:255 Avg:20
09:53:46 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:1 Avg:0
VOIP data from: 10.1.1.186:3000
09:53:56 Jitter:18 19 22 21 20 19 21 20 21 20: Buff:42 WC:255 Avg:20
09:53:56 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:1 Avg:0
VOIP data from: 10.1.1.186:3000
09:54:06 Jitter:19 22 20 18 18 18 18 19 18 18: Buff:30 WC:255 Avg:19
09:54:06 Pkloss:0 0 0 0 0 0 0 0 0 0: Oofo:0 WC:1 Avg:0

 

[montyzummer]

There is lots of jitter and packet lots happening here if you look at the * that means the packet loss is so intense the CM cant read that high ,also the jitter as you can see is going to be whats causing your echo.

Question is whats causing it , its obviously at the network level are you trying to use direct ip on your network region / sig groups ?

Are you using hairpinning , also what is your dsp resource medpro or G4XX gateway , as a first step i would try forcing the NR to use a dsp resource by turning off inter-region direct audio,

Trace it again and if you already have that setting off try using G729 (is this actually traversing a wan??)
??




09:53:26 Jitter:18 39 0 19 18 18 20 18 20 21: Buff:29 WC:255 Avg:21
09:53:26 Pkloss:0 0 * 0 0 0 0 0 0 0: Oofo:0 WC:0 Avg:0

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[telecomadmin12]

I got a medpro TN2302AP. Below I pasted network region and signaling group info. Anything I can change to improve this? I will try and use a different codec. The call was traversing a WAN. I will paste trace results.
Thanks a lot for your advise on this.
display signaling-group 1 Page 1 of 5
SIGNALING GROUP

Group Number: 1 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
IP Video? n Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 6
Supplementary Service Protocol: a
T303 Timer(sec): 10

Near-end Node Name: clan1 Far-end Node Name: samvaadabhumi
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 1
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n
Media Encryption? n Bypass If IP Threshold Exceeded? n
H.235 Annex H Required? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress
DCP/Analog Bearer Capability: 3.1kHz

display ip-network-region 1 Page 1 of 19
IP NETWORK REGION
Region: 1
Location: Authoritative Domain:
Name:
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: no
Codec Set: 1 Inter-region IP-IP Direct Audio: no
UDP Port Min: 2048 IP Audio Hairpinning? y
UDP Port Max: 3029
DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y
Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46 Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5

 

[montyzummer]

Sorry apologies i missed that you said you are using a medpro earlier.

Is your Asterix not in a separate NR ??

You could turn try to turn off hairpinning on your NR but that will mean more dsp resources are utilized , also add g729 to your codec set 1 , but this could well be your askerix config.

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ACSS (SME)
ACIS (UC)

 

[telecomadmin12]

My Asterisk is on the same LAN as my Avaya servers. Not sure how this relates to NR, does this mean they are on the same NR too?
I am also not sure what I am supposed tp put in the far end region field of my signaling group.
Strangely I can't make calls with G.729 codec although both Avaya and Asterisk support it.

 

[montyzummer]

You are saying the systems are on the same lan but traversing a wan , thats not possible.

You assign ip ranges in the ip-network-map and then tell that range what network region its in,

But you are traversing a wan so you should have separate regions.

Change ip-codec set 1 and add g729 , you can turn on silence suppresion off /on on your ip codec set.

Add another network region add another codec set and in your new network region form for the asterix tell that NR to use your new codec set i would have g729 in there only. so for NR 2 (your new one)the destination region 1 would use codec set 2 and force calls to be g729


Ideally you would have 3 regions using different codec sets , with an intervening region to manage bandwith between sites.

But you need to make sure the askerix is only using g729 also.

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[telecomadmin12]

My Avaya servers and my asterisk are on the same Lan. I was calling an analog Avaya extension from a remote sip client on asterisk (traversing wan). The sip client is out on the Internet. Avaya and Asterisk on the same Lan connecting via H.323. So I don't need to add another NR, right? Trying to figure out how to force Asterisk to use g729. And you think the problem is the codec?
And thanks again for the advice, I really appreciate it. This is a great site.

 

[montyzummer]

g729 will use less bandwith , but now i have a bigger picture about the sip client there is now a router probably firewall etc , so can you just get a simple h323 or sip device local on the lan connected to the asterix call and see if you get the same issue , you need to work out if this is lan or wan related my guess is wan ,

APSS (SME)
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ACIS (UC)

 

[telecomadmin12]

I don't get the same issue with calls local to the network. So the issue is related to WAN.

 

[montyzummer]

ok so we get somewhere great.... so explain exactly what the process is from start device each hop ip what the hop is and we may be able to narrow it down but as a first if you are using sip external turn off sip alg on the router.....a very very wide guess but at least from here we can narrow everthing down

APSS (SME)
ACSS (SME)
ACIS (UC)

 

[telecomadmin12]

Unfortunately the router that my ISP forces me to use is a very simplistic device and I don't see where I would be able to turn off sip alg.
But looks like I was able to resolve the issue for now by simply adjusting the mic settings on my remote client. Although the jitter issue persists.
I will surely post on other problems I have with these remote clients calling to analog in Avaya soon.

Thanks again for your help.

 

[montyzummer]

no probs glad you got somewhere

APSS (SME)
ACSS (SME)
ACIS (UC)