topic Re: Call is disconnected with SIP error 488 in Collaboration Applications

Call is disconnected with SIP error 488

Sebastien Denoncin — Tue, 15 Dec 2020 14:20:54 GMT

Hello the Community

I am currently experiencing a call disconnection problem between Jabber.

Call flow:

Caller Jabber --> CUCM --> UCCX --> CTI Port --> My Agent Jabber

When I check the SIP tarces, I found that we receive an SIP Message 488 Not Acceptable Media.

I think that it's a codec issue or something like that.

How can I confirm the root cause and How can I solve it?

Thanks for your help

Regards

Sébastien

Re: Call is disconnected with SIP error 488

nagaraj789 — Fri, 18 Dec 2020 07:55:15 GMT

You need to share the logs to determine the cause for failure. BTW, it might looks codec issue. If that is the case, You can try to use a transcoder and apply to SIP trunk device pool or make sure the audio codec preference list which you have set on dial-peers have all codecs listed such as G729br8,G729 etc.

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Fri, 18 Dec 2020 09:51:09 GMT

Hi Nagaraj,

The problem is that in my case we have not SIP trunk.

I noted that the problem occurs when the CSF is under recording. When CSF is not recorded, we have no problem.

I will clear my log and I will share it with you.

Thanks

regards

Sébastien

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Fri, 18 Dec 2020 13:38:55 GMT

Can check the codec preference in the gateway

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 isac mode independent bit-rate 32000 framesize 30
codec preference 3 g729r8

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Fri, 18 Dec 2020 13:39:33 GMT

Can share the sip traces this dicussion

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Fri, 18 Dec 2020 13:44:04 GMT

From SIP message are receive this message

Sent:

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP IP:5060;branch=z9hG4bK-b8555e-d00d7801-18f2e55

From: <sip:4033131776@ip>;tag=8e28930-0-13c4-55013-b8555b-71de78c1-b8555b

To: <sip:+12024655568@lii01.livun.com>;tag=56E55DB8-AC

Date: Tue, 17 Oct 2017 21:30:16 GMT

Call-ID: 31E69A60-B2B911E7-B7F6EA97-D6A08AF@lii01.livun.com

CSeq: 1 INVITE

Allow-Events: kpml, telephone-event

Server: Cisco-SIPGateway/IOS-15.5.3.S2

Content-Length: 0

Re: Call is disconnected with SIP error 488

Roger Kallberg — Fri, 18 Dec 2020 17:35:54 GMT

You would need to share a little bit more of the output of the debug than this.

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Sun, 20 Dec 2020 14:02:21 GMT

This sample 488 error message log next page

Supported: 100rel

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

Recv-Info: x-broadworks-client-session-info

Accept: application/btbc-session-info,application/media_control+xml,application/sdp,multipart/mixed

Max-Forwards: 19

Content-Type: application/sdp

Content-Length: 167

v=0

o=- 3305155623 1 IN IP4 172.30.1.4

s=-

c=IN IP4 172.30.1.4

t=0 0

m=audio 50194 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

076454: *Oct 17 2017 21:30:13.381: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=4033131776, Called Number=4033131776, Peer Info Type=DIALPEER_INFO_SPEECH

076455: *Oct 17 2017 21:30:13.381: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=4033131776

076456: *Oct 17 2017 21:30:13.381: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Sun, 20 Dec 2020 14:07:04 GMT

488 Not Acceptable Here
The response has the same meaning as 606 (Not Acceptable), but only
applies to the specific resource addressed by the Request-URI and the
request may succeed elsewhere.
A message body containing a description of media capabilities MAY be
present in the response, which is formatted according to the Accept
header field in the INVITE (or application/sdp if not present), the
same as a message body in a 200 (OK) response to an OPTIONS request.

Often this is related to codec incompatibilities. For anyone encountering this issue, they should check whether both sides (server and client) have at least one codes they can negotiate.

For fix that you should check both side configuration, I mean if you got an error for a SIP trunk you should check the SIP trunk provider codec and config if it’s interconnection in your company you should check the second server configuration.

For finding and debugging the codec for both side, you can capture SIP session by tools like Wireshark; you can check the codec for each call.

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Sun, 20 Dec 2020 14:07:58 GMT

problem is still unresolved, there is one more step.
Most of SIP provider want Early Offer INVITEs. They use this always to decide on which codec to offer for the calls.

Re: Call is disconnected with SIP error 488

Roger Kallberg — Sun, 20 Dec 2020 16:49:59 GMT

m=audio 50194 RTP/AVP 0 101

This line states what codec and DTMF method that is supported. It only list one codec, 0=g711ulaw, 101 is the payload type for DTMF.

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Wed, 23 Dec 2020 08:42:58 GMT

Hi Balaji,

Thanks for your reply.

But in my case, we don't go through the gateway.

It's for the internal call between Jabber for VDI and Contact Center Agent

Regards

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Wed, 23 Dec 2020 08:52:43 GMT

Hello the community,

I added the SDL trace from my CUCM sub for the impacted call.

Timestamp for the impacted call:

call starts: 10h28:32

End call:10h28:35

CallId: 0205857f-eb800007-000076d0-00002876

Thanks

Regards

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Thu, 24 Dec 2020 07:46:56 GMT

Hello,

After some tests, I noted that the service parameter G.722 codec enabled is not apply to Jabber Softphone.

In my case this parameter is disabled and when I do a test with one CSF, I can see that the invite contains G.722 and G.722.1 in SDP.

is it normal?

How can I remove this codec during the Invite?

Thanks

regards

Sébastien

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Thu, 24 Dec 2020 11:46:43 GMT

Codec preference can preferred from following link

https://www.google.com/search?q=how+to+change+the+codes+preferance+in+CUCM&oq=how+to+change+the+codes+preferance+in+CUCM&aqs=chrome..69i57.14118j1j15&sourceid=chrome&ie=UTF-8#kpvalbx=_GH_kX7uhM_3Az7sPnJ-mcA13

Re: Call is disconnected with SIP error 488

BalajiSivaraj49175 — Thu, 24 Dec 2020 11:47:11 GMT

SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP 10.99.206.223:60602;branch=z9hG4bK00002704
From: "Maude MIRTO" <sip:202505@ip>;tag=0205857feb8000370000296f-000021d7
To: <sip:6333@ip>;tag=27932613~b9ff6d1f-c80b-4887-8318-8f6b207ea5ac-42132316
Date: Tue, 15 Dec 2020 09:28:35 GMT
Call-ID: 0205857f-eb800007-000076d0-00002876@10.99.206.223
CSeq: 103 INVITE
Allow-Events: presence
Server: Cisco-CUCM11.5
Remote-Party-ID: "Maxime PAULIGNAN" <sip:975202451@ip>;party=called;screen=yes;privacy=off
Contact: <sip:6333@ip:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com="CSFBI14Y";video;bfcp
Content-Length: 0

39021291.004 |10:28:35.047 |AppInfo |EnvProcessCdr::outputCdrData CDR data (size-714)- 1,1,605892,42132316,1608024512,2,0,-540122358,"202505","BI194",0,47,4,-540122358,17348,40,20,0,0,0,0,0,0,"0","0",42132577,2,0,1942971146,"975202451","975202451","BI14Y",0,47,4,1942971146,17858,40,20,0,0,0,0,0,0,"0","0",1608024512,1608024515,"920114","878d306b-e116-46df-9005-47d9b0681454","p_iwe_global_internal","p_iwe_global_internal","p_iwe_global_internal","p_iwe_global_ccx",3,"CSFBI194","CSFBI14Y",13,13,0,10,0,4,0,"UCCluster",10,"","",0,"",3,3,0,0,64,64,"","","","10.99.206.223","10.99.207.115",103,3936,10,-540122358,32282,0,103,3936,10,1942971146,26036,0,0,"",3,"0000000000093EC40282E46100000000",0,0,0,"","",2,"","","","","","","","",0,0,"","","","","","","","",0,0,0,"975202451","975202451","920114",""

39021307.000 |10:28:35.047 |SdlSig |SIPDisconnReq |wait |SIPHandler(2,100,82,1) |SIPCdpc(2,100,85,635444) |2,100,14,339388.216^10.99.206.223^* |[T:N-H:0,N:0,L:0,V:0,Z:0,D:0] CcbId= 27932613 --TransType=1 --TransSecurity=0 PeerAddr = 10.99.206.223:60602 ccCause= 47 sip_disc_cause= 0
39021307.001 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 (SIP_APPLICATION_MSG), for event 8 (SIPSPI_EV_CC_CALL_DISCONNECT)
39021307.002 |10:28:35.047 |AppInfo |//SIP/Stack/States/0x0xd6209998/sipSPIChangeState: 0xd6209998 : State change from (STATE_MIDCALL_LOCAL_RESP_PENDING, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE)
39021307.003 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0xd6209998/sipSPIStopHoldTimer: Stopping hold timer
39021307.004 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0xd6209998/sipSPIPushOrigRequestContainerIntoHolde: Request Container Holder is above threshold...tri
39021307.005 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0xd6209998/sipSPIStopHoldTimer: Stopping hold timer
39021307.006 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0xd6209998/sipSPIAddViaParams: inserted z9hG4bK6b5aa43c5798c7 into via branch list
39021307.007 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0xd6209998/sipSPISendBye: Associated container=0xd5950fc0 to Bye
39021307.008 |10:28:35.047 |AppInfo |//SIP/Stack/Transport/0x0xd6209998/sipSPISendBye: Sending BYE to the transport layer
39021307.009 |10:28:35.047 |AppInfo |//SIP/Stack/Transport/0x0xd6209998/sipSPITransportSendMessage: msg=0xde1c4360, addr=10.99.206.223, port=60602, sentBy_port=0, is_req=1, tran
39021307.010 |10:28:35.047 |AppInfo |//SIP/Stack/Transport/0x0/sipInstanceGetConnectionId: gcb=0xd6209998 is already on connection=0xd6580e10 context_list
39021307.011 |10:28:35.047 |AppInfo |//SIP/Stack/Transport/0x0xd6209998/sipTransportLogicSendMsg: Set to send the msg=0xde1c4360
39021307.012 |10:28:35.047 |AppInfo |//SIP/Stack/Transport/0x0xd6209998/sipTransportPostSendMessage: Posting send for msg=0xde1c4360, addr=10.99.206.223, port=60602, connId=84391 for
39021307.013 |10:28:35.047 |AppInfo |//SIP/Stack/Info/0x0xd6209998/sentByeDisconnecting: Sent Bye Request, starting DisconnectTimer
39021307.014 |10:28:35.047 |AppInfo |//SIP/Stack/States/0x0xd6209998/sipSPIChangeState: 0xd6209998 : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
39021307.015 |10:28:35.047 |AppInfo |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: timerContext=0xd620a978 type=SIP_TIMER_DISCONNECT value=500 retries=10
39021307.016 |10:28:35.047 |AppInfo |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: timerContext=0xd620a978 type=SIP_TIMER_DISCONNECT value=500 retries=10
39021308.000 |10:28:35.048 |SdlSig |DChangeNotifyReq |restart0 |SIPBuiltInBridgeControl(2,100,86,42530) |SIPStationD(2,100,76,68029) |2,100,14,339388.216^10.99.206.223^* |[R:N-H:0,N:3,L:0,V:0,Z:0,D:0] CallPhase=CALL_CLEARING StationEvent=0 CI=42132577 EventQualifier=0
39021308.001 |10:28:35.048 |AppInfo |SIPBuiltInBridgeControl - getBibPrimaryCallData_byRefCI: no primary call with refCI=42132577
39021308.002 |10:28:35.048 |AppInfo |SIPBuiltInBridgeControl - getBibSessionCallData: no session with CI=42132577
39021308.003 |10:28:35.048 |AppInfo |SIPBuiltInBridgeControl::star_DChangeNotifyReq no call record for CI=42132577

Error SIP message

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Tue, 05 Jan 2021 09:25:56 GMT

Hello BalajiSivaraj49175

I already checked this step. All G711 are in the top of my list and G722 are in the bottom of my list.

But In the Invite, I can see the G722. I don't know why.

Thanks

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Tue, 05 Jan 2021 10:08:06 GMT

Hi BalajiSivaraj49175

Yes it's the SIP error message but I don't understand why we receive it.

Thanks

Regards

Sébastien

Re: Call is disconnected with SIP error 488

Roger Kallberg — Tue, 05 Jan 2021 13:49:35 GMT

You need to check why you don’t get a successful codec negotiation. The error indicates that there is either no codec agreed or a needed media resource is not able to be allocated for the call to succeed.

Re: Call is disconnected with SIP error 488

Sebastien Denoncin — Wed, 06 Jan 2021 13:19:47 GMT

Hello Roger,

I agree with you.

But I don't understand why the codec G722 and G722.1 are in the codec negociation. Because normally, G722 is disable on the CUCM Cluster and in the SIP profil, Cisco gave me a script to remove the G722.1 for another problem.

S normally both codec does not be used in the SIP invite with SDP. Right?

Thanks

regards

Sébastien