Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. And see here for current known issues.

New Member

Sip to H323 Call Flow.

Hello,

Does somebody knows where I can find:

- SIP to H323 call flow.

- SIP to H323 fax call flow.

Thanks in advance.

20 REPLIES

Re: Sip to H323 Call Flow.

I don't believe there are published anywhere.

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

-nick

New Member

Re: Sip to H323 Call Flow.

Hello,

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

Regards.

Re: Sip to H323 Call Flow.

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.

New Member

Re: Sip to H323 Call Flow.

Thanks tcatlinins.

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

Regards.

Re: Sip to H323 Call Flow.

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

Re: Sip to H323 Call Flow.

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

New Member

Re: Sip to H323 Call Flow.

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.

Re: Sip to H323 Call Flow.

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.

New Member

Re: Sip to H323 Call Flow.

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.

Re: Sip to H323 Call Flow.

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

New Member

Re: Sip to H323 Call Flow.

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.

Re: Sip to H323 Call Flow.

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

New Member

Re: Sip to H323 Call Flow.

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.

Re: Sip to H323 Call Flow.

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

New Member

Re: Sip to H323 Call Flow.

Hello,

For a new Invite with a new codec I should need a new Setup. How it would look the FAX call flow??

Thanks

Re: Sip to H323 Call Flow.

Actually for things like I listed - hold, resume, fax switchover, session refresh, etc you do not need a new setup for fax.

When we receive a RE-INVITE with T38 parameters in SIP, we will send an H323 H245 RequestMode out, and vice versa.

-nick

New Member

Re: Sip to H323 Call Flow.

Nicmath,

I'm attaching a successfull Fax Call flow. I can see the RE-INVITE with T38 parameters followed by an H323 H245 RequestMode.

What I still don't understand are the purpose of the TCS messages with T38 info, sent at the beggining of the call (see inside the red rectangle).

I'll appreciate your help.

Regards.

Re: Sip to H323 Call Flow.

This is one of the differences between SIP and H323.

For H323 all the session capabilities have to be exchanged at the beginning to know what is possible for that call.

In SIP, a RE-INVITE can renegotiate the capabilities, even if they were not negotiated to begin with.

If you were to disable fast start on the h323 side, you might see the capabilities sent to the other side.

hth,

nick

New Member

Re: Sip to H323 Call Flow.

Very helpful,

But, how can the SBC send a T.38 TCS message to the H323 Endpoint as a response to the first Facility TCS, if it hasn't received any T.38 info from the SIP UA???

H.323 ---(T.38 TCS)--->SBC

H.323 <--(T.38 TCS)----SBC ?????

Regards.

Re: Sip to H323 Call Flow.

If the SBC has T38 fax configured for the dial peer that was hit on the H323 leg, that is what matters. The media used for each leg will be determined by the dial peer unless you're using codec transparent. Even with codec transparent you'll likely see the fax capabilities.

-nick

1188
Views
10
Helpful
20
Replies
CreatePlease to create content