Sip to H323 Call Flow.

[guillermohurtado]

Hello,

Does somebody knows where I can find:

- SIP to H323 call flow.

- SIP to H323 fax call flow.

Thanks in advance.

[Nicholas Matthews]

I don't believe there are published anywhere.

Your best bet is to lab it up, if you have the ability.

-nick

[guillermohurtado]

Hello,

I was hoping someone had one already. I'll try to lab it up to post it later.

Regards.

[Tommer Catlin]

Well, here is one example:

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

sip

then on your dial peers

(example)

dial-peer voice 1 voip

translation-profile incoming voip

destination-pattern 4160

session protocol sipv2

session target ipv4:10.10.10.10

session transport tcp

incoming called-number +1415727....

codec g711ulaw

clid strip pi-restrict

And thats it on IOS. Just make sure you are using a later release of IOS and T train. The earlier ones are pretty buggy in 12.4.

[guillermohurtado]

Thanks tcatlinins.

But, I was looking for a graphical call flow between a SIP UA and a H323 Phone using, maybe, a SBC.

Regards.

[Tommer Catlin]

check out this link. Maybe this will help. It talks about ATT flex trunk and the Border Element usage.

Cheers

http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html

[Nicholas Matthews]

It is going to look something like this:

-->INVITE -->SETUP

<--100 Trying<--Proceeding

<--Alerting

<--180 Ringing

..ringing..

<--Connect

<--TCS

<--MSD

-->TCS

-->MSD

<--TCSAck

<--MSDack

-->TCSAck

-->MSDack

<--OLC

-->OLC

<--OLCack

-->OLCack

<-- 200 OK

--> ACK

.. call established..

--> BYE

<-- 200 OK

-->Release

<--Release complete

Brief, but should help. Important part to remember is the media from one side is negotiated before it is forwarded to the other.

-nick

[guillermohurtado]

Nicmatth,

I set up this scenario in lab, and I got the same you have posted. But,

- Why is there no communication to the SIP UA after the TCS and MSD? I was expecting a re-invite to the SIP UA with SDP description... Something like:

..ringing..

<--Connect

<--TCS

<--MSD

<--Invite SDP (t38)

-->Trying

-->200 ACK SDP (t38)

-->TCS

-->MSD

<--TCSAck

<--MSDack

-->TCSAck

-->MSDack

<--OLC

-->OLC

<--OLCack

-->OLCack

<-- 200 OK

--> ACK

Thanks a lot for your help.

Regards.

[Tommer Catlin]

Can you elaborate a little more? If you are using an IOS router, it's all done on the router, there really is nothing special to it other than using the dial-peer commands for setting options with H323 and SIP.

[guillermohurtado]

Nicmatth,

I set up this scenario in lab, and I got the same you have posted. But,

- Why is there no communication to the SIP UA after the TCS and MSD? I was expecting a re-invite to the SIP UA with SDP description... Something like:

..ringing..

<--Connect

<--TCS

<--MSD

<--Invite SDP (t38)

-->Trying

-->200 ACK SDP (t38)

-->TCS

-->MSD

<--TCSAck

<--MSDack

-->TCSAck

-->MSDack

<--OLC

-->OLC

<--OLCack

-->OLCack

<-- 200 OK

--> ACK

Thanks a lot for your help.

Regards.

[Nicholas Matthews]

You were expecting a re-invite in the middle of a call setup? For each SETUP we should have a single invite.

If you get into more complex call flows, things like hold/resume/faxing also will cause reinvites.

Since CUBE is a B2BUA it will make sure the media is entirely finished on one side before sending to the other. This means the full H245 negotiation will occur before we send a 200 OK with SDP.

On the flip side if we receive a 200 OK, we will do the H245 before sending an ACK.

Does that answer the question you were looking for?

-nick

[guillermohurtado]

Nicmath,

I'm attaching a fax call flow between a SIP UA and a H323 Endpoint through a SBC.

What I dont understand is the second Facility TCS (the one from the SIP UA to the H.323 EP). WHEN the SIP UA tells the SBC that it supports T38 ??.

Hope you understand.

Regards.

[Nicholas Matthews]

This is a call flow for H323 fast start. It looks normal to me.

It is slightly different than the slow start one that I outlined above, for simplicity's sake.

Even in fast-start H323, there will still be a TCS negotiation so things like DTMF and fax capabilities can be negotiated.

I don't see a problem here, unless you're wanting fax to work.

If so, you probably need to add:

voice service voip

fax protocol t38 fallback none

-nick

[guillermohurtado]

I understand there will be a TCS negotiation, but what I dont understand is WHEN or WHICH SIP message tells the H323 side that the SIP UA supports T38 fax? Shouldnt be a 200 OK w/SDP message right after receiving the TCS from the called side?

Regards.

[Nicholas Matthews]

I see what you're saying.

In a Re-Invite situation, the previous media is no longer applicable. The new invite could have a new codec in it that was not negotiated in the initial INVITE, since it is a new request.

As such, the SIP SBC shouldn't need to tell the other end about T38, because the RE-INVITE should be able to happen regardless.

-nick