CME - OOD Refer - INVITE/SDP Oddity

[danplacek]

I am trying to establish a call using an Out-Of-Dialog Refer... which works fine locally, but when I involve my SIP trunk, I get codec issues. For whatever reason "refer" calls do NOT set SDP in their outgoing INVITESs to the SIP trunk contrary to what the dial-peer used is set for (normal outgoing calls do). The provider defaults to G729, so the 2nd half of the refer call fails.

My REFER:

REFER sip:9<PSTN NUMBER>@192.168.10.1:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.18:5060;branch=asdf

From: <sip:999@192.168.10.18>;tag=a-tag

To: <sip:9<PSTN NUMBER>@192.168.10.1>

Call-ID: asdf4addaad3211

CSeq: 1 REFER

Refer-To: <sip:<PSTN NUMBER 2>@10.1.1.1>

Contact: <sip:192.168.10.18:5060>

Content-Length: 0

Which gets:

SIP/2.0 202 Accepted

Via: SIP/2.0/UDP 192.168.10.18:5060;branch=asdf;received=192.168.10.17

From: <sip:999@192.168.10.18>;tag=a-tag

To: <sip:9<PSTN NUMBER>@192.168.10.1>;tag=23C84774-F91

Date: Fri, 10 May 2013 04:51:00 GMT

Call-ID: asdf4addaad3211

CSeq: 1 REFER

Content-Length: 0

Contact: <sip:9<PSTN NUMBER>@192.168.10.1:5060>

This is the INVITE CME initiates for the first call:

INVITE sip:<PSTN NUMBER>@sip.flowroute.com:5060 SIP/2.0

Via: SIP/2.0/UDP <IP>:5060;branch=z9hG4bK19CB1F80

From: <sip:<CID>@sip.flowroute.com>;tag=23C84788-192

To: <sip:<PSTN NUMBER>@sip.flowroute.com>

Date: Fri, 10 May 2013 04:51:00 GMT

Call-ID: B8503D1-B86411E2-B12FA31B-D595BC53@108.81.15.25

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 0193146560-3093565922-2972492571-3583360083

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1368161460

Contact: <sip:<CID>@<MY IP>:5060>

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

Why no SDP????

After that call is answered, it immediately drops, and I see:

002331: May 10 04:51:09.956: //28684/0B832EC0B12C/SIP/Error/sipSPIDoAudioNegotiation: Media negotiation failed for m-line 1

002332: May 10 04:51:09.956: //28684/0B832EC0B12C/SIP/Error/sipSPIDoMediaNegotiation:

no valid fax or audio streams

002333: May 10 04:51:09.956: //28684/0B832EC0B12C/SIP/Error/sipSPICompareRespMediaInfo: Media Negotiation failed with no 18x Media

002334: May 10 04:51:09.956: //28684/0B832EC0B12C/SIP/Error/ccsip_api_call_connect_media: Media Info Failure

002335: May 10 04:51:09.956: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!!

002336: May 10 04:51:09.956: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!!

002337: May 10 04:51:09.956: //28684/0B832EC0B12C/CCAPI/cc_api_call_disconnected:

   Cause Value=65, Interface=0x87E94FDC, Call Id=28684

(and yes, the "200 OK" from the provider forces G729 since no preference was given in the INVITE)

This is the dial-peer used:

dial-peer voice 1020 voip

corlist outgoing call-national

translation-profile outgoing PSTN_Outgoing

preference 1

destination-pattern 91[2-9]..[2-9]......

session protocol sipv2

session target sip-server

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

codec g711ulaw

ip qos dscp cs5 media

ip qos dscp cs4 signaling

no vad

When dialing normally from a phone... it sends SDP with the INVITE.

Any suggestions appreciated... I cannot make heads or tails of this.

-Dan

[Ayodeji Okanlawon]

Daniel,

Thats a bit strange because by default CME sends early offer for calls. Maybe you should try and force EO under the sip-ua config

conf t

sip-ua

early-offer forced

Try that and see if the INVITE for the REFER includes SDP

Please rate all useful posts

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

[danplacek]

Thanks for the suggestion... but it appears that command is only available on CUBE? (or maybe newer CME, I am on 8.6)

LAB(config)#sip-ua

LAB(config-sip-ua)#early-offer forced

% Invalid input detected at '^' marker.

Any other suggestions?

Thanks,

Dan

[Ayodeji Okanlawon]

Daniel sorry its under sip config..

conf t

sip

early-offer forced

Please rate all useful posts

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

[danplacek]

I assume you mean `voice service voip` -> `sip` -- which worked.

However, I see the same issue, the INVITE has no SDP payload.

Any other thoughts?

Thanks,

Dan

[Ayodeji Okanlawon]

Dan..I am sorry I dont know why..What are u using the refer for?

Please rate all useful posts

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

[danplacek]

Click-to-dial and/or dial-through-office functionality.

sending an OOD Refer like that is an easy way to establish a call on behalf of a phone.

(or create a hairpinned call so that you can use your office number from your cell)

Obviously there are other ways to accomplish both of these goals, but the simplicity of a single REFER SIP message is attractive. It works flawlessly except for this media issue.

Thanks for your help so far.

-Dan

EDIT: to clarify, if it wasn't clear:

1. Send REFER with "To" set to your deskphone number or your cell

2. Set "Refer-to" header to number you want to call

3. Router will call your desk or cell

4. When you answer, it will start ring-out to the refer-to number.

This works with no issues internally. When involving PSTN numbers, it works, but has the media issues.

I imagine this would not be a problem if my provider defaulted to ulaw... but a lot of providers favor 729 for the bandwidth savings...