RE-INVITE responding with 488 Not Acceptable Media with IP-IP Gateway

Unanswered Question
Mar 8th, 2010
User Badges:
  • Silver, 250 points or more

I have been troubleshooting a problem with a IP to IP gateway, which is running IOS 12.4(20)T4.


Here is the relevant configuration (not with the real phone numbers or the real ITSP settings):

voice service voip
media flow-around
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
sip
  asserted-id pai
  rel1xx disable
  header-passing error-passthru
  registrar server
  midcall-signaling passthru
  sip-profiles 101
!
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
!
!
!
!
voice class sip-profiles 100
request INVITE sip-header P-Asserted-Identity modify "<sip:011(.*)" "<sip:+\1"
request INVITE sip-header Diversion modify "<sip:011(.*)" "<sip:+\1"
request INVITE sip-header P-Asserted-Identity modify "<sip:\+?(.*)" "<sip:+\1"
request INVITE sip-header Diversion modify "<sip:\+?(.*)" "<sip:+\1"
request REINVITE sip-header P-Asserted-Identity modify "<sip:011(.*)" "<sip:+\1"
request REINVITE sip-header Diversion modify "<sip:011(.*)" "<sip:+\1"
request REINVITE sip-header P-Asserted-Identity modify "<sip:\+?(.*)" "<sip:+\1"
request REINVITE sip-header Diversion modify "<sip:\+?(.*)" "<sip:+\1"
request INVITE sip-header Allow-Header modify " UPDATE, " " "
request REINVITE sip-header Allow-Header modify " UPDATE, " " "
response 180 sip-header Allow-Header modify " UPDATE, " " "
response 200 sip-header Allow-Header modify " UPDATE, " " "
!
voice class sip-profiles 101
request INVITE sip-header Allow-Header modify " UPDATE, " " "
request REINVITE sip-header Allow-Header modify " UPDATE, " " "
response 180 sip-header Allow-Header modify " UPDATE, " " "
response 200 sip-header Allow-Header modify " UPDATE, " " "
!
!
!
voice register global
mode cme
source-address 200.70.220.243 port 5060
max-dn 500
max-pool 185
authenticate register
authenticate realm ipipgw.example.com
!
voice register dn 1
number 14695552000
no-reg
!
voice register dn 2
number 14695552001
no-reg
!
voice register dn 3
number 14695552002
no-reg
!
voice register dn 4
number 14695554000
no-reg
!
voice register dn 5
number 14695554001
no-reg
!
voice register dn 6
number 14695554002
no-reg
!
voice register pool  1
id mac 001C.5555.3201
number 1 dn 1
number 2 dn 2
number 3 dn 3
dtmf-relay rtp-nte
voice-class codec 1
username 14695552000 password site1
no vad
!
voice register pool  2
id mac 001C.5555.9023
number 1 dn 4
number 2 dn 5
number 3 dn 6
dtmf-relay rtp-nte
voice-class codec 1
username 14695554000 password site2
no vad
!
!
voice translation-rule 1
rule 1 /^\+1/ /1/
rule 2 /^\+\([2-9]\)/ /011\1/
rule 3 /^\([2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9]\)$/ /1\1/
!
voice translation-rule 2
rule 1 /^011\([2-9]\)/ /+\1/
!
voice translation-profile Incoming_PSTN_Call
translate calling 1
translate called 1
translate redirect-target 1
translate redirect-called 1
!
voice translation-profile Outgoing_PSTN_Call
translate calling 2
!
dial-peer voice 1000 voip
translation-profile incoming Incoming_PSTN_Call
voice-class codec 1
session protocol sipv2
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 65 voip
translation-profile outgoing Outgoing_PSTN_Call
destination-pattern .%
voice-class codec 1
voice-class sip rel1xx disable
voice-class sip profiles 100
session protocol sipv2
session target dns:sipgateway.exampleitsp.com
session transport udp
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
!
!
!
sip-ua
no remote-party-id
!
!
!
gatekeeper
shutdown
!


I am able to make phone calls with the above configuration. I can also successfully forward calls with the above configuration. Not all calls fail to transfer properly. But on some phone calls, a 488 Not Acceptable Error response is returned instead of sending the re-INVITE to the ITSP gateway.


Here is what I really want to accomplish:

  • Media flow-around for calls
  • Re-INVITE messages sent to ITSP gateway


I had made a few changes, and I will try to see if rebooting it will solve the problem. Are there other settings that I should apply, or should I upgrade to a newer version of IOS?

  • 1
  • 2
  • 3
  • 4
  • 5
Overall Rating: 4.5 (2 ratings)
Loading.
John Platts Mon, 03/08/2010 - 21:54
User Badges:
  • Silver, 250 points or more

I actually executed the debug ccsip messages command on the IP-IP gateway. Here is the output of the re-INVITE, and the failed response:

Mar  8 23:53:20.691 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 76.250.30.1:5060;branch=z9hG4bK79C25ED
From: ;tag=5B031CC-1164
To: ;tag=2ACB58-1E1C
Date: Tue, 09 Mar 2010 05:43:33 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3481966201-0712380895-2157836914-4007938894
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: 1268113413
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 250

v=0
o=CiscoSystemsSIP-GW-UserAgent 5465 8927 IN IP4 76.250.30.1
s=SIP Call
c=IN IP4 76.250.30.1
t=0 0
m=audio 18542 RTP/AVP 0 101
c=IN IP4 76.250.30.1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Mar  8 23:53:20.691 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 76.250.30.1:5060;branch=z9hG4bK79C25ED
From: ;tag=5B031CC-1164
To: ;tag=2ACB58-1E1C
Date: Tue, 09 Mar 2010 05:53:20 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Mar  8 23:53:20.695 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 76.250.30.1:5060;branch=z9hG4bK79C25ED
From: ;tag=5B031CC-1164
To: ;tag=2ACB58-1E1C
Date: Tue, 09 Mar 2010 05:53:20 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

How do I get the IP-IP gateway to actually send the re-INVITE out to the ITSP SIP Gateway instead of sending back the 488 Not Acceptable Media? All three endpoints actually support the G.711 codec, and I still want media flow-around enabled. I do not want to transcode this call. How do I get this problem fixed? This re-INVITE is being handled incorrectly as we are getting a 488 response instead of the response being sent out to the ITSP SIP Gateway.

Steven Holl Wed, 09/29/2010 - 12:42
User Badges:
  • Cisco Employee,

For issues like this, run 'debug voip ccapi inout' along with 'debug ccsip mess' and see what inbound dial-peer is being matched.  Usually the inbound dial-peer match doesn't match the codec of which is being offered in the INVITE.

Actions

This Discussion

Related Content