Bandwidth for VoIP H323 signaling channels

Hallo
I have a VoIp application between two LANs connected by FR WAN.
I use two cisco1751 routers as gateway.
I would use LLQ and other QoS features on cisco gateway to prioritise VoIP streams.
access-list 100 permit udp host 192.168.1.1 host 192.168.2.2 range 16384 32767
access-list 101 permit tcp host 192.168.1.1 host 192.168.2.2 eq 1720
class-map voip-media
match access-group 100
exit
class-map voip-signaling
match access-group 101
exit
policy-map qos-voip
class voip-media
priority 12 (* number of VoIP channels)
class voip-signaling
bandwidth XXX
class class-default
fair-queue
Using g.729a codec and vad.....I would reserve 12kbps for each VoIp media channel (RTP) in the strict priority queue.
How many bandwith must I give to VoIP signaling (RTCP, H225) for each voIp channel ?
thankyou

Tejas:
I have a similar problem, I have voice over IP running over frame relay between a 7500 & a 3600 router. Doing DTS on the 7500 side. Everything works fine but when I try to use rtp compression to save some bandwith..it seems like "even though the command for rtp header-compresion are accepted on the interface configuration" Compression is actually not happening. Monitoring through EMCOM..sindle takes about 32K..i am suspecting that due to DTS on 7500 side..I am missing something....
Is there any good url ..you can suggest me....As far as enabling rtp header compression is concerned it is very straight forward...the requirement is that ditsributed CEF be enabled globally & interface level which I am already doing...
regards

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We have a customer who has subscribed to 1M of EF ( Expedite forwarding) class bandwidth ( in a 2M link)for transporting voice, AF1 class 392K for Voice signalling. The codec used is G.729b.The problem faced by the customer is they are not able to fully utilise the 1M bandwidth allocated for voice. The number of voice calls doesn't exceed 30 although there is enough bandwidth for about 35+ calls.There are also no drops seen in EF class.
Any idea whats causing this? Any specific debug which can be captured on the router when call volume peaks?
Many Thanks
ED

'show isdn call-rate' on some platforms will show you if the call flow is going through a PRI.
Otherwise there isn't much in the CLI that will tell you this, and you would need VOIP monitoring software.
'show voice call stat', 'show voice call summary', and 'show call active voice brief' will allow you to see many active calls are up at any given time, but no historical data.
You may want to try switching from G.729b to G.729. G.729b is going to have a varying bandwidth since it only sends packets when there is speech. If you use G.729 you will have a constant-rate codec rather than variable and you will receive better voice quality and more reliable QoS/bandwidth requirements.
-nick

Can I divide my Bandwidth dedicated for VoIP

Hi All,
Please advice me what I wanted to do is possible or not.
I have 128 kbps Bandwidth leased line circuit to my branch office in Singapore and I would like to dedicated assign 64 kbps of my bandwidth for VoIP traffice. The rest traffic with go with another 64 kbps.
I am using cisco 1760 routers and is it possible to do?
Any advice will be appreciated.

Bevilacqua,
I just configure on two Routers. Once I finised the configuration. I saw below output with not packets transfering.
Router#sh policy-map interface
Serial0/1
Service-policy output: llq
Class-map: voice (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: dscp ef
Queueing
Strict Priority
Output Queue: Conversation 40
Bandwidth 32 (kbps) Burst 800 (Bytes)
(pkts matched/bytes matched) 0/0
(total drops/bytes drops) 0/0
Class-map: class-default (match-any)
3269 packets, 2187341 bytes
5 minute offered rate 57000 bps, drop rate 0 bps
Match: any
Router#
After I made some ph calls. I can see below output showing packets transfering.
Router#sh policy-map interface
Serial0/1
Service-policy output: llq
Class-map: voice (match-all)
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5 minute offered rate 5000 bps, drop rate 0 bps
Match: dscp ef
Queueing
Strict Priority
Output Queue: Conversation 40
Bandwidth 32 (kbps) Burst 800 (Bytes)
(pkts matched/bytes matched) 3115/128688
(total drops/bytes drops) 0/0
Class-map: class-default (match-any)
16895 packets, 10045657 bytes
5 minute offered rate 81000 bps, drop rate 0 bps
Match: any
Router#
So, is it working fine now? How can I check it is actually reserver the bandwidth for voice?
Thanks for your kind help.

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Hi,
I have MPLS VPN scenario. For MPLS VPN at PE router connecting to hub, I've configured below QoS:
class-map match-any P1
match ip dscp ef
class-map match-any P3
match ip dscp default
class-map match-any P2
match ip dscp af31
policy-map QoS
class P1
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Since on the egree PE towords the internet, I'm using subinterface thus, the configuration is:
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class class-default
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service-policy output QoS
interface FastEthernet1/0.8
encapsulation dot1q 8
ip address ...
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Since the internet access is 256K, I've configured shape average 256000.
My problem is that, during high congestion at internet access, I noticed that on cust-hub my VoIP SIP is having problem to ring to another VoIP SIP within the same hub. Is it because of the shape command ?
thanks in advanced

Hi Maher,
shape average 256000 is to much. The point is, that you have OSI layer2 overhead and layer2 keepalives and such. Lower it to f.e. 240000. You might need to "play" a little with the parameter, as the overhaed depends on average packet size.
Also make sure, that your SIP traffic is marked with af31. Just as a note: Cisco changed from marking voip signaling with af31 to cs3. Take a sniffer and check that proper markings are in place for voip signaling. Or just try
class-map match-any P2
match ip dscp af31 cs3
Hope this helps! Please rate all posts.
Regards, Martin

Limit Data Bandwidth on VoIP

We are an US based international call centre.We are using cisco AS5300 as our VoIP gateway. I would like to know if i can use the same AS5300 for internet as well. If yes how can i limit the bandwidth out of the total bandwidth i have for data i.e. internet.Please let me know at the earliest.

Hello,
as an additional info to the previous post: I would recommend to use LLQ for voip traffic.
Not only a certain amount of bandwidth but also low delay are important for voip traffic.
So outbound you could use a policy like this:
ip cef
class-map match-any VoIP
match ip dscp ef cs5
match protocol rtp audio
class-map match-any VoIPsignal
match ip dscp cs3
policy-map VoIPprio
class VoIP
priority percent 10
class VoIPsignal
bandwidth remaining percent 5
class class-default
fair-queue
random-detect
interface Serial0
ip address ...
service-policy output VoIPprio
You would have to adjust the bandwith ratios for your needs and the interface naming to your environment.
The policy will give 10% of the link to voip traffic with lowest possible delay, 5% of the rest for voip signaling, and the rest for the other data present.
Again as already mentioned your ISP controls the interface towards you and you might need his helps to solve potential voip qos issues arising from dropped packets on his side.
Hope this helps! PLease rate all posts.
Regards, Martin

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Tips for dealing with large channel count on cRio

Hello, I have a very simple application that takes an analog input using an AI module (9205) from a thermistor and based on the value of the input it sends out a true/false signal using a digital out module (9477). Each cRio chassis will have close to 128 channels, the code being exactly the same for each channel.
I wonder if anyone has any tips for how I can do this so that I don't have to copy and paste each section of code 128 times. Obviously this would be a nightmare if the code ever had to be changed. I'm sure there is a way to make a function or a class but being new to graphical programming I can't think of a good way to do this. I looked for a way to dynamically select a channel but can't seem to find anything, if I could select the channel dynamically I'm guessing I can create a subvi and do it that way. Any tips or help would be greatly appreciated.

There isn't a way to dynamically choose a channel at runtime. In order
for the VI to compile successfully, the compiler must be able to statically
determine which channel is being read or written in the I/O Node at
compile time. However, that doesn't mean you can't write a reusable
subvi. If you right click on the FPGA I/O In terminal of the I/O Node and create a constant or control, you should be able to reuse the same logic for all of your channels. The attached screen shot should illustrate the basics of what this might look like. If you right click the I/O control/constant and select "Configure I/O Type...", you can configure the interface the I/O Item must support in order for it to be selectable from the control. While this helps single source some of the logic, you will still eventually need 128 I/O constants somewhere in your FPGA VI hierarchy.
I should also mention that if each channel being read from the 9205 is contained in a separate subVI or I/O Node, you will also incur some execution time overhead due to the scanning nature of the module. You mentioned you are reading temperature signals so the additional execution time may not be that important to you. If it is, you may want to look at the IO Sample Method. You can find more information and examples on how to use this method in the LV help. Using the IO Sample Method does allow you to dynamically choose a channel at runtime and is generally more efficient for high channel counts. However, it's also a lot more complicated to use than the I/O Node.
You also mentioned concerns about the size of arrays and the performance implications of using a single for loop to iterate across your data set. That's the classic design trade off when dealing with FPGAs. If you want to perform as much in parallel as possible, you'll need to store all 128 data points from the 9205 modules at once, process the data in parallel using 128 instances of the same circuit, and then output a digital value based on the result. If you're using fixed point data types, that's 182 x 26 bits for just the I/O data from the 9205. While this will yield the fastest execution times, the resulting VI may be too large to fit on your target. Conversely, you could use the IO Sample Method to read each channel one at a time, process the data using the same circuit, and then output a digital value. This strategy will use the least amount of logic on the FPGA but will also take the longest to execute. Of course, there are all sorts of options you could create in between these two extremes. Without knowing more about your requirements, it's hard to advise which end of the spectrum you should shoot for. Anyway, hopefully this will give you some ideas on where to get started.
Attachments:
IO Constant.JPG ‏31 KB

As5350 VOIP h323 g723 codec problem

I setup the VOIP network with H323. All calls are Gatekeeper routed.
When I make a PSTN call either inbound or outbound, as5350 could not
establish a call other than g729 codec.
Question 1. Why AS5350 uses g729 codec as default ? How you setup a
g729 default codec?
Question 2. Why AS5350 cannot use other codec like g723? AS5350
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h323
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pri-group timeslots 1-24
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progress_ind setup enable 3
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h323-gateway voip id AAAA ipaddr x.x.x.x 1719
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Bandwidth for VoIP H323 signaling channels

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