05-09-2013 10:08 PM - edited 03-16-2019 05:15 PM
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
05-10-2013 01:49 AM
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
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
05-10-2013 09:11 AM
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
05-10-2013 09:15 AM
Daniel sorry its under sip config..
conf t
sip
early-offer forced
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
05-10-2013 09:18 AM
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
05-10-2013 11:17 AM
Dan..I am sorry I dont know why..What are u using the refer for?
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"opportunity is a haughty goddess who waste no time with those who are unprepared"
05-10-2013 12:00 PM
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...
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