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Not able to call external conference number call gets disconnected aftera while

navshriv
Level 1
Level 1

Hi,

We seem to be able to recreate an issue with a specific called party where our CUBE SBC sends a 502 Bad Gateway message, this in turns causes the called party to disconnect the call. Can you please assist with resolving this fault?

The issue appears when calling 8567 7688 (in the examples from 9296 3596). After a short period of time, the call is disconnected.

In cucm traces  I can see after  the call is established  there is reinvite from cucbe

\\INVITE sip:294703596@10.19.1.4:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.19.10:5060;branch=z9hG4bK2D92F317E9
Remote-Party-ID: <sip:85677688@172.31.19.10>;party=calling;screen=no;privacy=off
From: <sip:85677688@172.31.19.10>;tag=E47EAB8-535
To: "Mark holmes" <sip:294703596@10.19.1.4>;tag=62547611~6c98aa1d-2c84-4263-8c45-7fcd46f15615-75793903
Date: Mon, 16 May 2016 00:41:02 GMT
Call-ID: 341c9e80-73911cba-742e8-401130a@10.19.1.4
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0874290816-0000065536-0000340123-0067179274
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 108 INVITE
Max-Forwards: 70
Timestamp: 1463359262
Contact: <sip:85677688@172.31.19.10:5060>
Expires: 300
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 623

v=0
o=CiscoSystemsSIP-GW-UserAgent 4793 6145 IN IP4 172.31.19.10
s=SIP Call
c=IN IP4 172.31.19.10
t=0 0
m=audio 29488 RTP/AVP 8 18 101 19
c=IN IP4 172.31.19.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
m=application 29796 UDP/BFCP *
c=IN IP4 172.31.19.10
a=floorctrl:s-only
a=confid:27562
a=userid:52385
a=floorid:2 mstrm:12
m=application 29892 UDP/UDT/IX *
c=IN IP4 172.31.19.10
a=setup:actpass
a=ixmap:0
a=ixmap:1
m=application 30008 RTP/AVP 100
c=IN IP4 172.31.19.10
a=rtpmap:100 H224/4800

\\ Cucm filtering the application mlines  and sending video port as 0

\\INVITE sip:4b7796d6-2a7d-9e45-f505-483f01842ebe@172.22.120.97:54892;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.19.1.4:5060;branch=z9hG4bK1bdd317fa2684f
From: <sip:085677688@10.19.1.4>;tag=62547564~6c98aa1d-2c84-4263-8c45-7fcd46f15615-75793902
To: "Mark Holmes" <sip:3596@10.19.1.4>;tag=0050b6c656dd00b9000016c5-00002390
Date: Mon, 16 May 2016 01:05:15 GMT
Call-ID: 0050b6c6-56dd0015-0000119f-00002c95@172.22.120.97
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 109 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; gci= 4-4402263; call-instance= 1
Remote-Party-ID: <sip:85677688@10.19.1.4>;party=calling;screen=no;privacy=off
Contact: <sip:3596@10.19.1.4:5060;transport=tcp>;video;audio
Content-Type: application/sdp
Content-Length: 431

v=0
o=CiscoSystemsCCM-SIP 2000 2 IN IP4 10.19.1.4
s=SIP Call
c=IN IP4 172.31.19.10
b=TIAS:64000
b=AS:64
t=0 0
m=audio 29488 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 119
b=TIAS:320000
a=rtpmap:119 H264/90000

\\but jabber replys  with two c attributes with video  port 0

SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.19.1.4:5060;branch=z9hG4bK1bdd317fa2684f
From: <sip:085677688@10.19.1.4>;tag=62547564~6c98aa1d-2c84-4263-8c45-7fcd46f15615-75793902
To: "Mark Holmes" <sip:3596@10.19.1.4>;tag=0050b6c656dd00b9000016c5-00002390
Call-ID: 0050b6c6-56dd0015-0000119f-00002c95@172.22.120.97
Date: Mon, 16 May 2016 01:05:16 GMT
CSeq: 109 INVITE
Server: Cisco-CSF/9.4.1
Contact: <sip:4b7796d6-2a7d-9e45-f505-483f01842ebe@172.22.120.97:54892;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Mark Holmes" <sip:3596@10.19.1.4>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 259
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 15205 1 IN IP4 172.22.120.97
s=SIP Call
b=AS:4000
t=0 0
m=audio 24206 RTP/AVP 8 101
c=IN IP4 172.22.120.97
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 0 RTP/AVP 119
c=IN IP4 0.0.0.0

\\ now cucm sends 200 ok to cube but without any sdp

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.19.10:5060;branch=z9hG4bK2D92F317E9
From: <sip:85677688@172.31.19.10>;tag=E47EAB8-535
To: "Mark holmes" <sip:294703596@10.19.1.4>;tag=62547611~6c98aa1d-2c84-4263-8c45-7fcd46f15615-75793903
Date: Mon, 16 May 2016 01:05:15 GMT
Call-ID: 341c9e80-73911cba-742e8-401130a@10.19.1.4
CSeq: 108 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: "Mark holmes" <sip:294703596@10.19.1.4>
Remote-Party-ID: "Mark holmes" <sip:294703596@10.19.1.4>;party=called;screen=yes;privacy=off
Contact: <sip:294703596@10.19.1.4:5060>;video
Content-Length: 0

\\So cube genrates bad gateway message to itsp  and call gets disconnected.

We utilise CUCM 8.5.1.12900-7 (10.19.1.4), which has a SIP trunk to our CUBE SBC (172.31.19.10), this has a sip trunk to our provider . Attached to  is the following information:
Capture.pcap - Packet capture of test call from the CUBE SBC.
Traces.zip - Trace files from CUCM of the test call from x3596 to 85677688 at 11:04am.
tech-support.txt - Output of show tech-support from the CUBE SBC.
debug-ccsip-all.txt - Output of debug ccsip all of a test call.

4 Replies 4

Tristan Cober
Level 1
Level 1

Looking over your logs I see this error after the 200 OK W/SDP from Jabber:

11:05:16.028 |//SIP/SIPHandler/ccbId=0/scbId=0/getVideoCodecFromSDP: rtpmap invalid for videoPayloadType=119|4,100,57,1.63592081^172.22.120.97^*
11:05:16.028 |//SIP/SIPHandler/ccbId=0/scbId=0/getVideoCodecFromSDP: Error! cannot convert pt=119|4,100,57,1.63592081^172.22.120.97^*

Your Jabber client doesn't like whats in the a=fmtp:119 for H.264, thus doesn't send an rtpmap, so the SDP fails to build out from CUCM.

I'm not 100% about how it should internally map the rtp to one it understands such as 97/126, but I'd imagine something in the H.264 capabilities is not supported on your Jabber client.

a=fmtp:119 profile-level-id=42E016;max-mbps=108000;max-fs=3600;max-dpb=5400

I would look into the Jabber logs, and send them off to TAC.

Edit: I assume if you disable video on the client configuration page in CUCM, the call would probably not drop.

Thanks for your  help  but the issue is we cant disable video on our network  we need that functionality to  our itsp. Also  we are having  the same issue calling from an ip phone I will get the  trace and attach.

As Tristan rightly pointed out that it looks like payload type mismatch.

You can try adding adding "asymmetric payload full "command either in global level or dial-peer level to see if it fixes the issue.

Global

sip

asymmetric payload full 

Dial-peer

dial-peer voice tag voip 4

voice-class sip asymmetric payload full

Hi,

having same issue.

above did not fix the issue.