Call Flow

[pakman]

Hello,

I want to learn how calls get setup logically from caller A to caller B if there on the same PBX, when there in different offices on different PBX's, and when caller A makes an outbond call. I understand how call routing works as I created some of this I want to learn more of the nuts and bolts but at a high level overview. Does anyone have any suggestions?To expand a little bit say caller A calls B in same building on same PBX what part of the call does the PBX take care of or does the PBX just setup the call and back out from there.

Thanks,

[LoopyLou]

Depends. If they are IP sets then the PBX is used to setup the call and then once the called party answers, voice in the form of RTP packets streams from the IP of the calling set to the IP of the called set. The PBX then just monitors the phones to see if either user presses a key ( like hold or transfer ) or whether either party hangs up. If the either set is analog then the PBX has to stay in the call as it does the conversion from IP to TDM.

If set A is on PBX A and set B on PBX B, then when set A calls sets B the PBX's have more involvement. Again depends if the phones are IP or analog. If set A is IP then it tells PBX it wants to talk to Set B. PBX A looks to see if the phone is local and if so the above applies. If it is not it has a lists of what PBX other sets are connected to. In this case it then sends a message to PBX B to rings set B. When set B answers it has been given the IP of Set A ( and visa versa ) and it streams RTP packets to the IP of Set A. Both PBX A and PBX B at this point are only monitoring.

In the case of an outside call, all the trunks ( lines ) are connected to the PBX. Therefore IP packets stream from the IP of the set to the IP of the PBX. In a 3300's case becuase the IP from the set needs to get converted to TDM before it can use other then a SIP trunk, then its the IP of the set to the IP of the E2T card.

Hopefully I haven't botched this up too much.

[pakman]

Thanks!

[LoopyLou]

Welcome. The big mistake people make is thinking that voice goes "through" the PBX controller for every call. Sure does for anything TDM but otherwise its IP to IP. Don't know how many times have had to trouble shoot bad or one way audio on IP phone to IP phone calls. Had to acutally unplug the controller from the network to show a customer that voice was not going throught the PBX on internal calls.

[pakman]

Thanks again...I do have another question. If phone A on PBX A is calling phone B on PBX B I understand that phone A's PBX looks up the phone and see's it's at another location. What I am fuzzy on does PBX A look to the switch for anything I wouldn't think so but wanted to check. In PBX A's programming I set the default gateway to the Router so i assume those packets just go out the router and the router sends those packets to PBX B could be more routers in between but for simplicity sake. Then once phone A and phone B are talking does anyone know how much bandwidth there taking up with compression? What would be the best tool to capture a test call in this scenario wireshark? I haven't used it that much but assume it could capture RTP packets?

Thanks

[LoopyLou]

In a cluster each PBX knows the IP address of every other PBX. So if PBX A needs to ring a phone and on PBX B it knows the IP it needs to send the message to. PBX A sends the message to the network and it gets routed by the network to PBX B. PBX A and B can be on the same VLAN or separated by a number of routers. Uncompressed its something like 100 K, compressed its 50K ( plus some overhead ). Best method of capturing a test call is as you suggest using Wireshark. That way you can watch the progression of the call from setup to speech.

Note but default voice is encrypted so you can't listen to the voice in a capture without it turned off.