End of an era..
This might be the last blog post for a little while (about 6 weeks) as I have passed my CCIE Voice and will now be taking a little well earned break.
Thanks everyone for reading, I really hope you got something out of this blog. I loved getting emails from you and comments, lots of people said nice things about some of the articles i wrote, that some hints and tips where useful to them. I am extremely glad that I could be of help to you.
I hope this blog helped someone out there!
Peter Revill
CCIE #18371 (Routing and Switching, Voice)
Conference Options in CME
Sorry its been a while since I posted, I have been diligently preparing for my CCIE Voice lab exam again (i'm going to smash it this time!)
So in tune with that, I have been studying lots of features in CME and CCM and I wanted to talk about conferencing in CME
First of all, heres a quick run down:
CME supports ad-hoc and meet-me conferencing.
Ad-hoc is supported in both software mode and hardware mode, while meet-me is only supported in hardware-mode conferences.
Hardware mode means that you must have a conference dspfarm resource registered to the CCME.
The ad-hoc in software mode is also limited to 3 participants as it uses a built in bridge on the phone. The participants must all be using the same codec.
Now here are some options for the software-based conferencing that you might not know:
First off, live-record (the cisco unity express feature) WILL work with software-based conferences as long as the codec used between all parties is G.711
Have you ever gone under an ephone and seen
"keep-conference"
and wondered what it was? well wonder no more, the keep-conference keyword is a way of controlling what happens when a conference initator leaves a software-based conference (it has no effect whatsoever on hardware-based conferences)
[no] keep-conference [end-call] [local-only]
Really quickly, heres how it works:
if no keep-conference is entered (the default) then if a conference initiator leaves the conference in any way the call is dropped, if the keep-conference command is there by itself, if the initator just hangs up the two other parties are connected to each other but if he presses 'end-call' then the call is dropped (strange I know)
if you have keep-conference end-call then no matter how the initiator hangs up the conference stays up. the local-only keyword ensures that all of these rules only apply if at least one local party is left in the conference.
So now you know the options around the software based conferences
hardware based conferencing (ad-hoc) adds the ability to add more than a certain number of participants (obviously) and also adds support for conflist (shows a list of conference participants) , rmlstc (removes the last caller) and join and select (allows joining of existing calls to conference)
the maximum participants for a conference is set under the dspfarm profile
dspfarm profile 1 conference
maximum conference-participants <>
!
now, some options available to you with conferencing on a hardware-based conference are:
conference add-mode creator
this command means that the only person allowed to add more people to a conference is the creator (this is done under the ephone) whenever that particular ephone creates a conference.
This might be useful for a CEO or someone who does not want other conference participants being able to add more people into the conference.
conference drop-mode creator and local
these are kind of the opposite way around to the keep-confernece command, by default a conference will always stay up if on a DSP farm even if the initiator leaves, with this command you can force it so that if the conference creator leaves the conference is dropped.
Now we come onto one more option that i think is very cool
conference admin
this command means that the phone tagged with this can admin any conference, so any conference he joins he can remove participants and add more participants, one very cool feature it has is that he can even join existing conferences by dialing the ad-hoc number of the conference!
I hope this helps someone out there
Advanced H323 Gatekeeper troubleshooting
Advanced Gatekeeper stuff/Troubleshooting
HI Guys,
So, first things first, how do we troubleshoot for example, when a remote gatekeeper has the bandwidth command restricting us from calling? Example config given below:
gatekeeper
zone local REMOTE cisco.com 113.29.243.26
zone local ZONE1 cisco.com
bandwidth remote 10
no shutdown
!
Lets say we try and send this guy a call, obviously it won’t work, since we need enough bandwidth for a call both ways, (128)
Here is what we will see when we do a show h323 gateway on the GW trying to make a call through our own local GK to this remote GK:
PSTN-WAN# show h323 gateway
….
DISC CAUSE CODE FROM OTHER PEER FROM H323 PEER
34 no circuit (34) 0 3
I tried to call three times here hence the 3 J
Now LOCALLY if our own LOCAL GK (or a GK that they tell us to point to but don’t actually let us go onto) has the bandwidth command, we get a much more useful message (arj)
gatekeeper
zone local MYZONE cisco.com
zone remote REMOTE cisco.com 113.29.243.26 1719
zone prefix REMOTE 6*
gw-type-prefix 3* hopoff MYZONE gw ipaddr 10.0.0.3 1720
bandwidth remote 10
no shutdown
endpoint resource-threshold
So lets check out what happens now, clear our stats again:
PSTN-WAN#clear h323 gateway
All H.323 stats cleared at 1w0d
<- make test call.. beep beep beep beep!->
PSTN-WAN#show h323 gateway ras
RAS STATISTICS AT 1w0d
RAS MESSAGE REQUESTS SENT CONFIRMS RCVD REJECTS RCVD
GK Discovery grq 0 gcf 0 grj 0
Registration rrq 0 rcf 0 rrj 0
Admission arq 1 acf 0 arj 1
You can see here we TRIED to make a call (arq) but where rejected (arj) probably due to the bandwidth issue.
This message will also appear if you do a “endpoint” restriction, heres an example config for this:
gatekeeper
zone local MYZONE cisco.com
zone remote REMOTE cisco.com 113.29.243.26 1719
zone prefix REMOTE 6*
gw-type-prefix 3* hopoff MYZONE gw ipaddr 10.0.0.3 1720
no shutdown
endpoint resource-threshold
endpoint max-calls h323id PSTN-WAN 1
!
You MUST have the “Endpoint resource-threshold” configured for this to actually work
Once again, if you have this configured, the message sent back if you try and make more than two calls will be admission reject
RAS MESSAGE REQUESTS SENT CONFIRMS RCVD REJECTS RCVD
GK Discovery grq 0 gcf 0 grj 0
Registration rrq 4 rcf 4 rrj 0
Admission arq 2 acf 0 arj 2
Bandwidth brq 0 bcf 0 brj 0
Disengage drq 0 dcf 0 drj 0
Unregister urq 0 ucf 0 urj 0
Resource Avail rai 0 rac 0
Req In Progress rip 0
QOS and the error message " CBWFQ : Not supported on subinterfaces"
Pete here, so I am doing some more labs, really focusing on IP-IP-GW interaction and QOS to try and get across the line with that, so here is the first thing I wanted to chat about
CBWFQ : Not supported on subinterfaces
Ever seen that error message before? it happens when you try and apply a QOS policy with either the priority command or the bandwidth command on an interface.
HOWEVER, what I did NOT know, is that you can use hierarchical policy maps when faced with this message, observe the below:
BR1-R2#show policy-map
Policy Map voip
Class voip
priority 20 (%)
Class class-default
fair-queue
Policy Map 512
Class class-default
Average Rate Traffic Shaping
cir 512000 (bps) bc 5120 (bits)
service-policy voip
if you now apply the policy-map 512:
interface Serial0/0/1:0.1 point-to-point
description == FR To HQ
ip address 177.0.101.2 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 101
service-policy output 512
end
You can see that it is accepting the fair-queue:
BR1-R2#show policy-map int serial 0/0/1:0.1
Serial0/0/1:0.1
Service-policy output: 512
Class-map: class-default (match-any)
102 packets, 9309 bytes
5 minute offered rate 1000 bps, drop rate 0 bps
Match: any
Queueing
queue limit 64 packets
(queue depth/total drops/no-buffer drops) 0/0/0
(pkts output/bytes output) 102/7795
shape (average) cir 512000, bc 5120, be 5120
target shape rate 512000
lower bound cir 0, adapt to fecn 0
Service-policy : voip
queue stats for all priority classes:
queue limit 64 packets
(queue depth/total drops/no-buffer drops) 0/0/0
(pkts output/bytes output) 0/0
Class-map: voip (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: protocol rtp
Priority: 20% (102 kbps), burst bytes 2550, b/w exceed drops: 0
Class-map: class-default (match-any)
102 packets, 9309 bytes
5 minute offered rate 1000 bps, drop rate 0 bps
Match: any
Queueing
queue limit 64 packets
(queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0
(pkts output/bytes output) 102/7795
Fair-queue: per-flow queue limit 16
I hope this helps someone out there!
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