OCS Sip Server Call Flow

[Gui75]

Hi All,

I installed OCS and Sip Server in my laborary.
I was surprised to see that sip call flow involved does not involve any REFER message or RE-INVITE.
This is not a transfer callflow.

Simply, there is first an INVITE without sdp towards Gateway which responds with SDP offer including RTP port and IP Adress and list of codecs

Second, an INVITE is sent to the agent with the sdp offer got from gateway . Agent SIP Phone answers with OK and SDP answer including RTP Port and IP Address of the sip phone

Third ACK is sent to the gateway with the previous sdp response which permits to get the rtp connection between agent and Gateway.


Well... Why not.
Anyone can confirm that this callflow is the good one?
Is there any means to change the call flow to send a REFER to the gateway?

Regards,

[René]

Hi,

Described call flow is correct as DN used by an agent is provisioned on SIP Server. SIP Server have to stay in signalling path to track call status. Using REFER would mean that SIP Server will be excluded from signalling path thus losing possibility to track the call.

Standard OCS call flows are described in the "Outbound Contact 7.x Reference Manual".

R.

[Gui75]

Thanks René fpr your answer.

I saw indeed that Standard Call flows are described in Outbound 7 reference manual. But they are described in a very light way without describing sip messages involved in it.

When you say that REFER excludes Sip Server from signalling path, I have some doubts, an option oosp-transfer-enabled was introduced to make it possible to bypass Sip Server but it is false by default. I think that putting Sip Server out of signalling path depends on what you put in Refer-To header : it can be something like 444@IP_Sip_Server. In this case refer should be handled by Sip Server right?

At least, I forgot to give another detail. In fact in my architecture, agents are located on a PBX behind the same gateway (doing multisite transfer to keep context). But I think that your statement remain : Sip Server must remain in the signalling path

Regards,

[René]

You're right about the call flows in the documentation. These are not very detailed in the meaning of SIP messages.

Option "oosp-transfer-enabled" allows you to take SIP Server out of signaling path using REFER message. Personally, I don't see any point using this option for single-step transfer and specifying SIP Server object in Refer-To. But maybe my imagination isn't good enough

I don't know the architecture of your system but in case the agents are located on PBX and not on SIP Server you can try to use REFER. SIP Server doesn't need to control the call as it is transferred to another Genesys-controlled (my assumption) switch. T-Server for that switch will take control over the call and communication with OCS (update of call result etc.). Does it make sense?

R.

[Gui75]

Thanks René,
Well Rene, regarding your assumption, I really do not know.

But I would like to check it.
What I do not know is what I have to configure to get such a call flow with REFER for OCS Outbound.

[René]

I've been thinking about that again and I'm not sure my yesterday post about possible usage of REFER is correct. The reason I've changed my mind is that we're not talking about single-step transfer but about call being routed to an agent.

Simple OCS call flow:
1/ OCS issues request to SIP Server
2/ SIP Server sends INVITE to gateway
3/ Call is established on selected Routing Point (Voice Transfer Destination) and routing strategy is executed by URS
4/ Target agent is found - URS issues RequestRouteCall to SIP Server
5/ SIP Server sends INVITE to agent's endpoint
6/ Call between external party and the agent is established

Key step is number five when SIP Server routes call to an agent. I think there isn't possibility to say SIP Server to use REFER as it is an outbound call.

What do you think about that?

R.

[Gui75]

Thanks Rene, I think EXACTLY the same as you.

Unfortunately, I did not try to convince anybody that I was right, despite I installed everything on my laptop and made it work.

So I will have to wait next implementation on customer to see if I am right or not.

It should be in 2 weeks.
I will keep you in touch.

Regards,