Meet-me Conference for SIP based IP phone

[Ajasco]

Hello,
I am trying to configure Meet-me conference for our network users. All our cisco IP phone 7940G have been converted to SIP based phone. It is stated that A sip-based phone can only join the meet-me conference but not create one.

Please help. Is this an accurate statement? Does that mean I would have to convert all the phones back to SCCP based phone?

Please help. Is there a way to do this both on the CME 4.1 and voice register pool configured for the SIP phone? Thanks all

Ajasco

 

[whykap]

This is from the 4.x admin guide:
SIP Functions Supported in Cisco CallManager:
Basic Outgoing Call
You can initiate outgoing calls to a SIP device from any Cisco CallManager device. A Cisco CallManager device includes SCCP IP Phones or fax devices that are connected to Foreign Exchange Station (FXS) gateways. For example, an SCCP IP Phone can place a call to a SIP endpoint. The SIP device answering the call triggers media establishment.

Basic Incoming Call
Any device on the SIP network, including SIP IP Phones or fax devices that are connected to FXS gateways can initiate incoming calls. For example, a SIP endpoint can initiate a call to an SCCP IP Phone. The SCCP IP Phone answering the call triggers media establishment.

Use of Early Media
While the PSTN provides inband progress information to signal early media (such as a ring tone or a busy signal), the same does not hold true for SIP. The originating party includes Session Description Protocol (SDP) information, such as codec usage, IP address, and port number, in the outgoing INVITE message. In response, the terminating party sends its codec, IP address, and port number in a 183 Session Progress message to indicate possible early media.

The 183 Session Progress response indicates that the message body contains information about the media session. Both 180 Alerting and 183 Session Progress messages may contain SDP, which allows an early media session to be established prior to the call being answered.

When early media needs to be delivered to SIP endpoints prior to connection, Cisco CallManager always sends a 183 Session Progress message with SDP. While Cisco CallManager does not generate a 180 Alerting message with SDP, it does support the 180 Alerting message with SDP when it receives one.

Supplementary Services Initiated by SIP Endpoint:
SIP-Initiated Call Transfer
Cisco CallManager does not support SIP-initiated call transfer and does not accept receiving REFER requests or INVITE messages that include a Replaces header. When Cisco CallManager receives a REFER request, it returns a 501 Not Implemented message. When Cisco CallManager receives an INVITE message with a Replaces header, it processes the call and ignores the Replaces header.

Call Hold
Cisco CallManager supports call hold and retrieve that a SIP device initiates or that a Cisco CallManager device initiates. For example, when a SCCP IP Phone user retrieves a call that was placed on hold by another user, Cisco CallManager sends a re-INVITE message to the SIP proxy. The re-INVITE message contains updated Remote-Party-ID information to reflect the current connected party. If Cisco CallManager originally initiated the call, the Party field in the Remote-Party-ID header gets set to calling; otherwise, it gets set to called. For more information on the Party field parameter, refer to Enhanced Call Identification Services.

Call Forward
Cisco CallManager supports call forward that a SIP device initiates or that a Cisco CallManager device initiates. With call forwarding redirection requests from SIP devices, Cisco CallManager processes the requests. For call forwarding initiated by Cisco CallManager, no SIP redirection messages are used. Cisco CallManager handles redirection internally then conveys the connected party information to the originating SIP endpoint through the Remote-Party-Id header.



So unless you upgrade to version 6.x where CISCO SIP register phones have all the functions that SCCP does you will have to convert them all back to sccp.

Out of curiosity. Why did you convert all your phones to SIP? What reason did you have? I certainly cannot think of one.

 

[Ajasco]

Thanks Whykap for the information.
I inherited this voip phone system and I have been working with it for almost a year. Not until a month ago I was tasked with meet-me conference.

We run SIP protocol internally and used H323 to communicate amongst the branches. If I change to sccp based phone, Doesn't that mean I won't be using SIP internally anymore? That's architectural change that will require a compelling reason.

If version 6 will allow me to do this meet-me conference, then I like to upgrade.

Please, tell me. Is version 6 integrated into 2800 IOS router?

Thanks

Ajasco

 

[whykap]

Sorry, I did not read your post carefully. the info above is for call manager and not CME.
I am not sure if you can do meet me on version 4.x CME with SIP phones. Let me research it and get back to you.

Sorry for the wrong information.