[cisco-voip] QS: TFTP Server and Phone Loads

[Syed Khalid Ali]

If I switch to G.729 for branch office will this make any difference.  I will also need a Transcoder as well... right? 

What about MOH? Will switching to G.729 has an advantages. I read in CIPT 4.x book that G.729 for MOH is designed to Latin Languages and for others it will sound like crap.... Any one having practical experience with such stuff, please share your experience.

Thanks
Khalid
CC: cisco-voip at puck.nether.net; thsglobal at gmail.com
From: rratliff at cisco.com
Subject: Re: [cisco-voip] QS: TFTP Server and Phone Loads
Date: Mon, 15 Sep 2008 09:27:13 -0400
To: khalid_khursheed at hotmail.com


I mean packet routing.   In order to get 2-way audio CUCM has to receive and send the RTP parameters (IP address, port, and codec) from each party and send them on to the other party.   After that it's all routing in the network as RTP is just another UDP data flow.   
 -Ryan  
On Sep 12, 2008, at 10:51 PM, Syed Khalid Ali wrote:Thanks for the reply and explanation.

Regarding the one-way audio, what do you mean by routing issue. It is call routing or packet network routing? I have other branches running and there are no issues at all but the media is either DSL, DXX or WiMAX and the delay is around 100-250 ms.

Note: I am runnning Call Manager 4.2(3) with IPCC Express 4.0(4)

Thanks
Khalid
CC: cisco-voip at puck.nether.net; thsglobal at gmail.com
From: rratliff at cisco.com
Subject: Re: [cisco-voip] QS: TFTP Server and Phone Loads
Date: Fri, 12 Sep 2008 14:02:43 -0400
To: khalid_khursheed at hotmail.com

Are your phones having to upgrade the phone firmware?
The zip file alone for the 7911 version 8.3.5 is about 4Mb.  If my quick and dirty math is correct at 512 bytes per packet it will take about 8000 packets to download that 4Mb of phone load files, per phone.  At 800ms per packet (assuming that's RTT so best-case) that's a very, very long time to upgrade phone loads.
I'd highly encourage you to get either a local TFTP server for phone loads (recent CUCM versions allow you to set a load server at the phone level) or set those phones to use whatever load they are currently on so they can skip the upgrading part.  
One-way audio is going to be a routing issue, the transport won't change that fact.  
-Ryan
On Sep 12, 2008, at 12:41 PM, Syed Khalid Ali wrote:Hi,
 
I have 7911G IP Phones. We have up to 4 to 5 phones at each site. I have tested it via ping command from call manager to ip phones. The delay with 32 bytes packet is around 650 to 675 ms. For 200 to 300 bytes packet the delay goes around 800+ms delay. The link capacity is 256k/256k. I did not measure the packet loss throughly, however, I did check show ip interface output and did not find any packet loss or CRC errors.

I have waited  register ofor around 1 hour just tone phone. Hell only one of the phone was able to register with CallManager. Also what about Peer Firmware Sharing with 7911 series?

Also for 3-4 phones, it will be very difficult to justify the cost for CME at remote site to administration.

One more thing, the problem of ONE-WAY Audio is also there!

Waiting for your response.

Regards,
Khalid

From: rratliff at cisco.com
Subject: Re: [cisco-voip] QS: TFTP Server and Phone Loads
Date: Fri, 12 Sep 2008 10:21:05 -0400
To: khalid_khursheed at hotmail.com

For phone loads it depends on he phone model and version.  The 7940/60 loads are pretty small, the 3rd-gen phone loads can get pretty big.  Over a high latency link you will see some very long times to upgrade 3rd gen phones as TFTP only transmits a small amount of data (512 bytes IIRC) and each packet has to be acked before the next is sent.
For registration the VSAT won't be too bad.  Delay itself won't even make voice quality too bad as long as the jitter is low.  Somebody who has actually used it may have more specific information though.
-Ryan
On Sep 12, 2008, at 2:29 AM, Syed Khalid Ali wrote:All,

How TFTP server provides phone load to ip phones. I mean what is the minimum number of BYTES tftp server sends to IP Phones. 

Another things is that we have some branches in location VSAT is the only possible solution. Will ip phones be able to register with call manager.

Using CCM 4.2(3).

Regards,
Khalid

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