SIP trunk early offer

[j.huizinga]

Hi,

We have configured a trunk with a provider using CUBE.

Callmanager--------CUBE--------Provider

The provider wants early offer and G729r8

So we configured a SIP trunk with a device pool/region so that only g729 is allowed between SIP trunk and the rest.

We have configured an IOS MTP resource, and this is registered on callmanager:

!

dspfarm profile 2 mtp

codec g729r8

maximum sessions software 20

associate application SCCP

!

!

The mtp resource is assign to the trunk using MR-list and MR-group

The trunk has MTP enabled with "MTP prefered Codec" G729b/G729ab

On CUBE we enable "deb ccsip mess" and we see the invite comming from callmanager, but without attached SDP

What must be done to make callmanager use early-offer?

Thanks for the help,

Jan

[Ayodeji Okanlawon]

Can you send the output of this command..

show sip-ua status

Please rate useful posts

"For the love of God is broader than the measure of man's mind And the heart of the Eternal is most wonderfully kind"

[j.huizinga]

Hi,

At this moment I have removed this router from customer site, and it is in my office.

We got crazy about this router not accepting the bind commands.

But regardless of configuration, the bind commands should be accepted I think.

Thanks,

Jan

[jfernandorosa]

Hi Aokanlawon,

I have the similar problem:

My scenario:

CUCM 9X -->SIP TRUNK--> CUBE --> ISP SIP

I need invite Early Offer to ISP, for DTMF problems, I don´t like the use CUCM fot this.

I set in CUBE (early-offer forced), but if I removed pass-thru content sdp, i received fas busy and CUCM return Internal Server Error:

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  early-offer forced

  midcall-signaling passthru

  pass-thru headers unsupp

  no call service stop

how can I solve this?

Thanks!

Joao

[Ayodeji Okanlawon]

Can you do a test call and send us "debug ccsip messages" attach it here

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[jfernandorosa]

Hi,

Attached Log in last post.

Thanks,

Joao

[Ayodeji Okanlawon]

The log you attached was a succesful call and your cube didnt send EO to your ITSP. I didnt see any error in the log

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[jfernandorosa]

In this case not have problem, is correct.

But in some calls the ISP don´t invite SDP payload with DTMF information (telephone event) and DTMF fails in this case.

I attached log problem.

Thanks.

Joao

[Ayodeji Okanlawon]

There is nothing you can do, if your ITSP doesnt advertise any DTMF capabilites in their SDP. You need to contact them and have it corrected. CUBE can only respond to what is offered. This is a problem with them so get them to sort it out

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[jfernandorosa]

Hi.

I remove the SIP configs below for invite SDP EO to ISP.

sip

  pass-thru headers unsupp

  pass-thru content sdp

  no call service stop

and use (early-offer forced)

The ISP response with payload complete in this case, I dialed the same number with problems, but my call rinring and return a fast busy in this case, CUBE return internal server error for CUCM.

Sent:

SIP/2.0 500 Internal Server Error

Via: SIP/2.0/TCP 21.10.0.7:5060;branch=z9hG4bK13eb293705d2

From: "ATA187 Core" <6001>;tag=21576~fb89236f-816b-47f5-8c94-b8d3c388dd7c-64665064

To: <297840042484>;tag=3E21E738-DD8

Date: Mon, 02 Sep 2013 21:44:24 GMT

Call-ID: 263d9280-22510742-b70-7000a15@21.10.0.7

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M3

Reason: Q.850;cause=96

Content-Length: 0

[Ayodeji Okanlawon]

You need to post the full debug, for us to know whats happening. Cause code 96 means that a madatory IE is missing

Typical scenarios include:

• Mandatory Contact field missing in SIP message.
• Session Description Protocol (SDP) body is missing.

So until I see the full log, I wont know what is wrong

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[jfernandorosa]

Hi,

Attached logs with error 96 for you analise.

I don´t see the SDP payloads.

Thanks for help.

Joao

[jbollen]

Well, in this case (error code 96) the service provider is:

a. removig the SDP in the subsequent 180 Ringing

b. not including an SDP in the 200 OK

That's why the router spits out cause code 96.

Talk to the Service Provider, I would suggest...

cheers,

Jan

[Mukesh Kumar]

Hi Ayodeji,

We have the following setup:

Phones-- PBX--- Voice gateway --- SIP provider 

When we try to make any outbound call, call gets connected and when the destination end receives the call the call gets disconnected. 

It shows the cause 16 of disconnecting the call.

As I have searched for it, I found that call is cleared normally. But we haven't cleared call.

When I have checked the debugs on Voice Gateway. I have found that when we make any outboud call then after registration, we receive 183 Session progress message from the SIP provider. After that SDP message send from our end.

It looks like that the SIP provider is using early offer and on our end delay offer is running.

Can you please tell me if one voice gateway is using delay offer and the SIP provider will be using early offer then what will happen?

Does the call be successful or it has got connected?

Regards;

MUKESH KUMAR | Network Engineer

Spooster IT Services

Computer Networking Solutions

mjindal@spooster.com

www.spooster.com

[Vivek Batra]

Which PBX?

Which voice gateway?

When I have checked the debugs on Voice Gateway. I have found that when we make any outboud call then after registration, we receive 183 Session progress message from the SIP provider. After that SDP message send from our end.

It looks like that the SIP provider is using early offer and on our end delay offer is running.

It's always UAC who can decide whether to use early or delayed offer. So in your case, when you make an outgoing call, it's your gateway who can choose between early and delayed offer, not your service provider. It's true vice-versa when gateway receives incoming call from service provider, your service provider decides whether to use early or delayed offer.

Coming back to your question and comments, we can't say it was early or delayed offer as you've not mentioned whether INVITE from gateway was sent with or without SDP. If it was with SDP, gateway is using early offer and if it was without SDP, gateway is using delayed offer.

Can you please tell me if one voice gateway is using delay offer and the SIP provider will be using early offer then what will happen?

The question is not much relevant because UAS has to respond as per request revived from UAC. If UAC has initiated call using delayed offer, UAS must have to support delayed offer else UAS should reject the call if doesn't support delayed offer.

[Mukesh Kumar]

Thanks for the information Vivek.

In our case call is disconnected from our end.

When the call gets connected and the destination receives the call then from our end "bye" is sent to the SIP provider.

Is this is related to SIP early offer or SIP early delay?

Calling number is on our end and the called number is on the SIP provider end.

Regards,

MUKESH KUMAR | Network Engineer

Spooster IT Services

Computer Networking Solutions

mjindal@spooster.com

www.spooster.com