Some basic questions..

[billy.davies]

Hello All,

First post here. We are about to look at rolling out a Cisco IP Telephony solution and are currently looking for partners to work with us..

Before we do that, I wanted to understand some basics if that's OK?

I understand that the call setup is achieved via RTP which is seperate to the the voice packets? 

I've read that RTP is not often sent in the EF class and is normally sent in the AF class - is this the case, if so, what kind of bandwidth do we need to allocate for RTP?

Looking at the models - Centralised, Distributed, Deployment and Geographical Diversity (I think that covers it).  Our model looks to be distributed with one large Campus site and lots of smaller branches - if someone in a branch calls someone else in the same branch, does the actual call go via the HQ site or is it just the RTP (i.e. call setup) that is sent to the HQ - i.e. are the actual voice packets are sent across the same LAN or do they have to be sent via the HQ?

Thanks in advance, William.

[Jaime Valencia]

RTP flow IS the voice, there is also signaling which is either SCCP or SIP.

RTP by default is tagged as EF.

Table 3-3     Traffic Classification Guidelines for Various Types of Network  Traffic 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1044066

QoS questions are covered in the same link, you might want to go thru it if you're new to this.

SRNDs are the bibles when it comes to design and understanding.

RTP unless there is a TRP/MTP need is sent directly between endpoints. Only signaling is always sent back to CUCM.

Unified Communications Deployment Models chapter covers the deployment models just in case you want to read some more about that.

HTH

java

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[billy.davies]

Hi, thanks for the response. So, if a user calls another user on the same LAN, would the voice packets go direct between the 2 users on the LAN? I'm assuming just the call setup is sent to the HQ but the actual packets are sent direct between users, is this correct?

Thanks, Wil,

[Jaime Valencia]

Correct, most IP phone to IP phone calls will look like this:

IP Phone 1 --- RTP --- IP Phone 2

Only the signaling for both endpoints is sent back to CUCM.

HTH

java

If this helps, please rate

www.cisco.com/go/pdihelpdesk

[billy.davies]

Great, and last question(s).. Do you know approx how much bandwidth we need to reserve for call setup and does the SIP call setup go via the AF class?

Lastly... We were recommended 40k a call over Ethernet WAN's, 27k a call over E1 - sound about right?

[Jonathan Schulenberg]

The SRND provides some figures on how much bandwidth is required for call signling (table 3-11) and voice media (table 3-9 & 10): http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/netstruc.html#wp1044815

Per-call bandwidth depends one what codec you use (e.g. G.711 vs. G.729) and what network transport it crosses. You'll need to determine the codec choice for intra- and inter-location calls. Additionally, http://www.bandcalc.com/ provides a handy tool for multiplying this out.