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Silver

SIP trunk early offer

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

Everyone's tags (1)
58 REPLIES
Cisco Employee

Re: SIP trunk early offer

Check MTP required on the SIP trunk configuration page or if you are on CUCM 8.5 or later go to the SIP Profile (Device > Device Settings > SIP Profile) and check "Early Offer support for voice and video calls (insert MTP if needed)".

Silver

Re: SIP trunk early offer

Hello Joe,

I have MTP required enabled on the trunk, but it doesn't really change anything.

If I enable, or don't enable "MTP required" the SIP INVITE from callmanager is the same.

We are running Callmanager 7.1.5, so we don't have the option in the SIP profile.

When we establish a call I can see that the IOS MTP is involved (show sccp connections) but no early offer.

To make a successfull cal we force the provider dial-peer to use G729 only.

Unfortunately this IOS doesn't have the forced early offer option.

Is there someone who has succeeded in making CM 7.1.5 do an early offer?

Thanks,

Jan

Cisco Employee

Re: SIP trunk early offer

Hi Jan,

Here is your answer

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/media.html#wp1055833

Snippet from above

SIP Early Offer

SIP negotiates media exchange via Session Description Protocol (SDP),  where one side offers a set of capabilities to which the other side  answers, thus converging on a set of media characteristics. SIP allows  the initial offer to be sent either by the caller in the initial INVITE  message or, if the caller chooses not to, the called party can send the  initial offer in the first reliable response. By default, Unified CM  sends the INVITE without an initial offer, and it requires MTP resources  to send the offer in the INVITE. Note that this initial offer is  limited to the G.711 codec only.

Also note that MTP resources are not required for incoming INVITE messages, whether or not they contain an initial offer.

Regards

Niket

Silver

Re: SIP trunk early offer

Hi Niket,

You seem to be right, when I force 711 I see early offer.

But it is weird that on the trunk configuration page, it allows me to select a g729 codec as prefered codec.

Can you explain why this option is there, although it doesn't sem to work?

Thanks for your help,

Jan

Cisco Employee

Re: SIP trunk early offer

Hi Jan,

I dont know whats the logic and i dont have a confirmed answer, behind not supporting the g729 codec in CM 7.1.5 , however when i open the SRND for CM 8.x

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1126420

I dont see any requirements on the codec.

So somewhere between CM 7.x and CM 8.x there must be a fix for supporting the g729 codec on early offer.

Hope this helps.

Regards

Niket

Silver

Re: SIP trunk early offer

Hi Niket,

Thank you very much.

We are testing now with another router that has an IOS which supports 'SIP early offer forced' and we hope that this will work.

So the callmanager will send a normal INVITE (delayed offer) and the CUBE will send out a early offer to the provider.

Regards,

Jan

VIP Super Bronze

SIP trunk early offer

J,

Did the early offer-forced work?

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Silver

SIP trunk early offer

Hi,

We haven't configured yet. I shall let the community know.

Thanks for all,

Jan

VIP Super Bronze

SIP trunk early offer

J,

I just saw your query on the ask the expert forum. I thought I should chip in here for you.

Yes CUCM can send delay offer and CUBE can send early offer. This is how we have it configured for one of my biggest customers with 3 clusters (8 server/cluster) and it works very well...

Here is a snippet from our CUBE config..

sip

early-offer forced.

Here is a sample trace..

+++++CUCM sends invite without early offer+++++++++

Received:
INVITE sip:07544455678@172.16.10.74:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.174:5060;branch=z9hG4bKa41e64e002602
From: "Chelmsford" <01256748210>;tag=517285~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-400694295
To: <07544455678>
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 67d44180-fde1f8d8-3aead-ae28690a@172.16.10.174
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <172.16.10.174:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1741963648-0000065536-0000064326-2921883914
Session-Expires:  84600
Contact: <01256748210>
Max-Forwards: 70
Content-Length: 0

++++++++++++++CUBE sends a trying to CUCM++++++++++++++++

Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.10.174:5060;branch=z9hG4bKa41e64e002602
From: "Chelmsford" <01256748210>;tag=517285~ffa80926-5fac-4dd6-b405-2dbbc56ae9a2-400694295
To: <07544455678>
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 67d44180-fde1f8d8-3aead-ae28690a@172.16.10.174
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

+++++++++CUBE sends an invite to ITSP with Early Offer++++++++++


010625: Jun 18 09:46:00.371: //452987/67D441800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:07544455678@10.100.33.54:5070 SIP/2.0
Via: SIP/2.0/UDP 172.16.10.74:5060;branch=z9hG4bKFE71F2421
Remote-Party-ID: "Chelmsford" <441256748210>;party=calling;screen=no;privacy=off
From: "Chelmsford" <441256748210>;tag=88EFEFBA-14A2
To: <07544455678>
Date: Mon, 18 Jun 2012 09:46:00 GMT
Call-ID: 3EDB1C4A-B86111E1-94E48F4D-5D7E5E41@172.16.10.74
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 1741963648-0000065536-0000064326-2921883914
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1340012760
Contact: <441256748210>
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires:  84600
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 354

v=0
o=CiscoSystemsSIP-GW-UserAgent 6030 604 IN IP4 172.16.10.74
s=SIP Call
c=IN IP4 172.16.10.74
t=0 0
m=audio 18110 RTP/AVP 18 0 8 100 101
c=IN IP4 172.16.10.74
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

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Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Silver

SIP trunk early offer

Thank you very much!

We are going to configure it like this.

Unfortunately one of our CUBE router has not the forced SIP early offer command, and we need to install more flash and memory and then do an IOS upgrade.

We now make a test with a 2811 that has the correct IOS, but this router has such a weird issue that I think I have to open a TAC case. In this router I can not apply the "bind control source-interface" command. I can type it, but the"show run" doesn't show it. So weird.

We did format flash, install another IOS, but it doesn't apply the bind commands.

Thanks!

Jan

VIP Super Bronze

SIP trunk early offer

How did you apply the command..Is it like this..

sip

  bind control source-interface gig0/0

  bind media source-interface gig0/0

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Silver

SIP trunk early offer

Exactly!

but in the show run we don't see it

Thanks,

Jan

VIP Super Bronze

SIP trunk early offer

What IOS version are you running?

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Silver

SIP trunk early offer

c2800nm-ipvoice_ivs-mz.124-20.T6.bin

c2800nm-adventerprisek9-mz.124-24.T7.bin

c2800nm-adventerprisek9-mz.124-24.T4.bin

c2800nm-adventerprisek9-mz.124-24.T5.bin

Pick one

We tried all of them.

Thanks,

Jan

VIP Super Bronze

SIP trunk early offer

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
Silver

SIP trunk early offer

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

New Member

Re: SIP trunk early offer

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

VIP Super Bronze

SIP trunk early offer

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

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Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP trunk early offer

Hi,

Attached Log in last post.

Thanks,

Joao

VIP Super Bronze

SIP trunk early offer

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP trunk early offer

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

VIP Super Bronze

SIP trunk early offer

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Re: SIP trunk early offer

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



VIP Super Bronze

SIP trunk early offer

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"

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

SIP trunk early offer

Hi,

Attached logs with error 96 for you analise.

I don´t see the SDP payloads.

Thanks for help.

Joao

New Member

Re: SIP trunk early offer

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

New Member

Hi Ayodeji,We have the

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

 

 

 

Which PBX?

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.

New Member

Thanks for the information

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
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