Sccp call setup diagram

Start on. Show related SlideShares at end. WordPress Shortcode. Share Email. Top clipped slide. Download Now Download Download to read offline. Call flows. Page Flow Diagram. Prioritizing handoffs.

Gsm architecture and call flow. Complete umts call flow. W level3 wcdma rno rf optimizationa-1[1]. Channel assignment strategies. Related Books Free with a 30 day trial from Scribd. Now to the end user may not be noticeable, unless that DNS server is down. If for some reason the DNS server is down or unavailable, now I can't do my name resolution, I can't find my Communications Manager server.

Like you can see on the next image I am able to from the IP phones perspective look up by the IP address where that communications server lives. So, I don't have to go through the extra step. In the Communications Manager I can go to system server and that's where I can statically configure the IP address of the server.

Right now by default if you just install the system there's a name there, so if I remove the name, put in the IP address, I have now removed the DNS reliance and what'll happen too is the TFTP files that my phone downloads will also contain the IP addresses so there really will be no DNS reliance whatsoever, once I've made these changes. Let's start to look at the call flows. Within my own environment if I have a centralized and a remote branch location what will happen is my signalling will take place.

I pick up the phone, I am using Skinny and I am dialing digits. The centralized environment says the Communications Manager is going to know where that remote phone is. So if I dial extension for example, I now do the look up via the Communications Manager and the Communications Manger says "Oh you're looking for a phone over in this branch location". So now the branch location is signalled ringing, hopefully they pick up the handset and answer the call and once that takes place the Communications Manager kind of steps out of the loop.

Now, it could be going through various routers and WAN-links to reach that destination, but the Communications Manager, steps out of the link and the media is going to flow through that path between your phones. If we have a centralized deployment, and I have an IP WAN Link maybe reaching out to my branch locations and for some reason it goes down — what are my options? Well, you might not have a backup link and then we wouldn't be able to communicate with our branch locations, but my guess is most environments you have a connection out to the PSTN and you would rather the call be placed even if you have a little bit of long-distance charge incurred through the public switch telephone network.

So what'll happen is if I pick up my handset, I dial my digits and the Communications Manager attempts to send the call down the IP WAN path and finds that "Oh wait a minute it's down, we could re-route the call through the public switch telephone network to that branch location". Now, with the branch location I just want to explain a few things that you're seeing on the image below. What that means is that branch router has special software running on it so that it could take over as the phone system.

So it's going to manage all of those phones in that branch location because remember we lost the WAN connection and if this is a centralized deployment I've now lost access to the Communications Manager which was managing those phones. So first off, SRST needs to be running on that branch router.

Example shows the Cisco CME router trying to set up a call to the number dialed As soon as the matching destination is found, the calling phone is updated with the destination's caller ID details. The called phone is set to ringing mode and is also updated with the caller ID of the calling phone. Example shows phone 2 answering the call.

When it does so, the Cisco CME router sends an "Open Receive Channel" message to both the phones, specifying the codec and packet size. The phones acknowledge this message by sending an "Open Receive Channel Ack" message specifying the port number to start the media transmission. You can see that ephone 1 and ephone 2 send the "Open Receive Channel Ack" message specifying the port numbers and , respectively.

As soon as Cisco CME knows the port numbers for both the phones, a "StartMedia" message is sent to the phones, specifying the port numbers of the other phones to send media to. At this point, the phones start sending media to each other. Example shows the call clearing. Phone 2 disconnects the call by going on-hook.

This tears down the media path between the two phones. Also, an on-hook message is sent to phone 1. The highlighted output shown in Examples to points out the key messages to look for when troubleshooting call flow issues. For example, if the problem is no dial tone when the phone goes off-hook, you should look for the "StartTone" message shown in Example that the Cisco CME router sends to the phone.

If the problem is that the call does not connect, you should look for the "OpenReceive" and "StartMedia" message exchanges shown in Example between the endpoints. The output of the debugs can be quite verbose.

Looking for the key messages can help you identify the problem easier. Figure shows a sample exchange of messages between an H. The exchange is very similar to the call setup discussed in the previous section except for the signaling messages specific to the H.

You can view the call setup and teardown messages by turning on the debug h asn1 and debug h asn1 commands. The output of these debug commands shows the contents of the incoming and outgoing H.