[cisco-voip] SIP Load and Re-Invite

Fri Apr 15 14:49:04 EDT 2011

Not sure what a CMG tone is but any codec change from CUCM is going to require a signaling request.  Prior to 8.5 CUCM required a SIP call to set the media inactive via SDP before changing codec mid-call.  This requirement has been removed with 8.5.

> For this release, Cisco Unified Communications Manager also enhances interoperability with third party devices during mid-call operations including basic hold/resume operations and during supplementary services, such as transfer and conference. In previous releases, Cisco Unified Communications Manager sent an INVITE with an inactive SDP (a=inactive attribute) to indicate a break in media path, sent a Delayed Offer INVITE to insert music on hold or resume the media stream, and expected a send-recv offer SDP in the 200 OK. Because third party devices often provide an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path would remain in an inactive state and cause calls to drop. In this release, Cisco Unified Communications Manager allows you to configure a parameter for an early offer SIP trunk so that Cisco Unified Communications Manager suppresses the sending of inactive or sendonly SDP in mid-call INVITEs. When this parameter gets enabled, Cisco Unified Communications Manager connects the SIP Trunk device directly to the MOH or annunciator device without breaking the existing media stream during call hold or other feature invocation. Similarly, Cisco Unified Communications Manager connects the SIP Trunk device to a line side device directly during call resume without breaking the MOH or annunciator stream. By preventing the far end media stream from getting set to inactive, Cisco Unified Communications Manager should always be able to resume the media path.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/8_5_1/delta/delta.html#wp1849900

For detecting tones and taking action you are going to have a DSP in the RTP stream that can intercept the tones and signal upstream the required action.   I'm not sure what the point is in a 79XX phone responding to something like a fax tone since it couldn't do anything with said fax data anyway.

Long story short, yes CUCM is completely capable of changing codecs mid-call, as long as you signal the codec change in a way we can actually act on.

-Ryan

On Apr 15, 2011, at 2:19 PM, Mike Lydick wrote:

The field test was Verizon introducing CMG tones a few seconds after the call was established to a Cisco 7931 phone. The Cube logged: "Failed to negotiate media: 488".

In our example we have CUBE and CM8.5 in place but the call fails to switch the Media stream codec after a CMG tone is sent.

We are not  using 'Require MTP', but tried the Dynamic MTP option in the SIP Profile on UCM but the call still fails after the codec change is sent and this option appears to alway enable the MTP.

Tac is questioning if this supported and what scenario would require this. The only one that I can think is an Analog port that has a multifunction Fax, or a line that is being shared for voice an modem. We are going to retest on a FXS port on Monday.

I see an option for modifying SDP for mid call codec changes, we did not enable at the time of testing. Do not have any documentation that states that this is required?

This SIP stuff is a fad anyways I am going to tell the customer to move back to H323...

Best Regards,

Mike Lydick

On Fri, Apr 15, 2011 at 1:35 PM, Dennis Heim <Dennis.Heim at cdw.com> wrote:
Do you have a cube in place?

Dennis Heim
Network Voice Engineer
CDW  Advanced Technology Services
11711 N. Meridian Street, Suite 225
Carmel, IN  46032

317.569.4255 Single Number Reach
317.569.4201 Fax 
dennis.heim at cdw.com
cdw.com/content/solutions/unified-communications/

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (AM)
Sent: Friday, April 15, 2011 12:51 PM
To: cisco-voip (cisco-voip at puck.nether.net)
Subject: [cisco-voip] SIP Load and Re-Invite

Is it normal for sip providers (say Verizon) to want to change codec mid-call or require your equipment can do it?  I understand CallManager 8.5 / SIP 7945 can’t do this.  Is it a CallManager limitation or a phone load limitation or both for reinvite to change codec mid-call?

I’m a fan of codecs that dynamically change bandwidth (Silk, iLBC) but not G.722 to G711 to G.729 if the call degrades, but I guess with SIP you could possible do this based on the spec for re-invite. Or switch from G.729 to G.711 for faxes, etc.

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