[cisco-voip] SIp Trunk call failing after PBX upgrade

Tue Mar 26 12:55:56 EDT 2019

Actually meant o= line is the origin line.

On Tue, Mar 26, 2019 at 12:39 PM Brian Meade <bmeade90 at vt.edu> wrote:

> It's definitely failing at parsing the SDP on that invite and finding an
> invalid parameter:
> 07517620.001 |16:00:23.657 |AppInfo  |//SIP/SIPUdp/wait_UdpDataInd:
> Incoming SIP UDP message size 932 from 172.27.0.15:[5060]:
> [1031135,NET]
> INVITE sip:3366 at 10.4.128.27 SIP/2.0
> Via: SIP/2.0/UDP 172.27.0.15:11347
> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
> From: "Gabriel Querol" <sip:86329 at 172.27.0.15>;tag=2792862
> To: <sip:3366 at 10.4.128.27>
> Call-ID: 501227892-15 at 172.27.0.15
> CSeq: 1 INVITE
> Contact: <sip:86329 at 172.27.0.15:11347;transport=udp>
> Max-Forwards: 70
> User-Agent: MitE1x v4.4.5.1062
> Expires: 300
> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
> P-Early-Media: Supported
> P-Asserted-Identity: "Gabriel Querol" <sip:86329 at 172.27.0.15>
> P-Mitrol-idLlamada: 190322160050689_MIT_07437
> P-Mitrol-LoginID: gquerol
> P-Mitrol-PerfilRuteo: 100
> Content-Length: 233
> Content-Type: application/sdp
>
> v=0
> o=86329 -835641967 1 IN IP4 172.27.0.15
> s=MitE1x Call
> c=IN IP4 172.27.0.15
> t=0 0
> m=audio 36112 RTP/AVP 0 8 101
> a=sendrecv
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 07517621.007 |16:00:23.657 |AppInfo
> |//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed -
> sdp_ret=SDP_INVALID_PARAMETER
>
> You may need to use a SIP Normalization script to clean up what they are
> sending.
>
> I think it's the o= line (organization line).  That's 2nd value
> (-835641967) should be a positive number I believe.  That session-id
> parameter is supposed to match NTP format-
> https://tools.ietf.org/html/rfc4566#section-5.2
>
> Maybe just check their server has NTP synced okay to start?
>
> Thanks,
> Brian Meade
>
>
>
> On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel <Ariel.ROZA at la.logicalis.com>
> wrote:
>
>> Here´s the trace file with the bad call
>>
>>
>>
>>
>>
>>
>>
>> *De:* Brian Meade <bmeade90 at vt.edu>
>> *Enviado el:* lunes, 25 de marzo de 2019 23:39
>> *Para:* ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM>
>> *CC:* Jonatan Quezada <jonatan.quezada at chemeketa.edu>; cisco-voip (
>> cisco-voip at puck.nether.net) <cisco-voip at puck.nether.net>
>> *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
>>
>>
>>
>> Can you send the trace file you pulled the bad call from?
>>
>>
>>
>> Is MTP Required set on the SIP Trunk?
>>
>>
>>
>> On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel <Ariel.ROZA at la.logicalis.com>
>> wrote:
>>
>> My issue is not a CUCM upgrade. The other side from the SIP Trunk was the
>> one that was updated (a local in-house development, called Mitrol). The
>> system worked fine before the upgrade, and after that it went bonkers.
>>
>>
>>
>> *De:* Jonatan Quezada <jonatan.quezada at chemeketa.edu>
>> *Enviado el:* lunes, 25 de marzo de 2019 19:24
>> *Para:* ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM>
>> *CC:* cisco-voip (cisco-voip at puck.nether.net) <cisco-voip at puck.nether.net
>> >
>> *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
>>
>>
>>
>> we are seeing a similar issues to one of our nodes. we did our during
>> production, Brave but totally doable. After figuring out that we needed to
>> point the EM profiles to the node we were keeping up for the upgrade, we
>> took down the other ucs down, all went well for upgrade. All VM on my ucs
>> are all done now, but there is this huge jitter issues that has risen from
>> the ashes of the upgrade. Its as if my media RTP streams are being forked
>> and the forking is causing the jitter and delay?
>>
>>
>>
>> I have calls where I lose second of audio but signaling seems fine, Im
>> just losing a ton of packets between the nodes now that they(the pub and
>> sub) are load balancing the media resources, or rather seeming to load
>> ballance.
>>
>>
>>
>> After some dial peer and server group re pointing, all devices finally
>> were on the one node and we were able to upgrade the UCS, but the other is
>> left to do. all of my CUCM
>>
>>
>>
>> On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel <Ariel.ROZA at la.logicalis.com>
>> wrote:
>>
>> Hi, guys and gals.
>>
>>
>>
>> I have a customer with a CUCM 9.0(2) cluster.
>>
>> It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or
>> otherwise). The PBX has four different nodes, all configured in the SIP
>> TRUNK
>>
>>
>>
>> They claim it was working fine until last Thursday, where they did an
>> upgrade to one of the nodes of the PBX. After that, calls going from PBX to
>> CUCM fail with a 488 Media Not Acceptable error.
>>
>> They also have tried making calls from one of the not upgraded nodes,
>> with the same error.
>>
>> I have been looking into the SIP traces, and I see nothing really telling
>> of a problem there.
>>
>>
>>
>> We reseted the SIP trunk with no success.
>>
>> I have looked at the región configuration, and all regions are set to the
>> System Default (G722, G711)
>>
>> I also tried changing the preferred codec in the SIP trunk, with no
>> success.
>>
>>
>>
>> Following this, I am pasting the SIP messages of a failed call from PBX
>> -> CUCM and a successfull call in the reverse, from CUCM -> PBX.
>>
>>
>>
>> Can you see if anything is wrong or odd?
>>
>>
>>
>> Regards,
>>
>>
>>
>> Ariel.
>>
>>
>>
>> Failed Call from PBX
>>
>> --------------------
>>
>>
>>
>> INVITE sip:3366 at 10.4.128.27 SIP/2.0
>>
>> Via: SIP/2.0/UDP 172.27.0.15:11347
>> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
>>
>> From: "XXXX XXXX" <sip:86329 at 172.27.0.15>;tag=2792862
>>
>> To: <sip:3366 at 10.4.128.27>
>>
>> Call-ID: 501227892-15 at 172.27.0.15
>>
>> CSeq: 1 INVITE
>>
>> Contact: <sip:86329 at 172.27.0.15:11347;transport=udp>
>>
>> Max-Forwards: 70
>>
>> User-Agent: MitE1x v4.4.5.1062
>>
>> Expires: 300
>>
>> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
>>
>> P-Early-Media: Supported
>>
>> P-Asserted-Identity: "XXXX XXXX" <sip:86329 at 172.27.0.15>
>>
>> P-Mitrol-idLlamada: 190322160050689_MIT_07437
>>
>> P-Mitrol-LoginID: XXXX
>>
>> P-Mitrol-PerfilRuteo: 100
>>
>> Content-Length: 233
>>
>> Content-Type: application/sdp
>>
>> v=0
>>
>> o=86329 -835641967 1 IN IP4 172.27.0.15
>>
>> s=MitE1x Call
>>
>> c=IN IP4 172.27.0.15
>>
>> t=0 0
>>
>> m=audio 36112 RTP/AVP 0 8 101
>>
>> a=sendrecv
>>
>> a=rtpmap:0 PCMU/8000/1
>>
>> a=rtpmap:8 PCMA/8000/1
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>>
>>
>>
>>
>> Reply from CUCM
>>
>> ---------------
>>
>>
>>
>> SIP/2.0 488 Not Acceptable Media
>>
>> Via: SIP/2.0/UDP 172.27.0.15:11347
>> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm
>>
>> From: "Gabriel Querol" <sip:86329 at 172.27.0.15>;tag=2792862
>>
>> To: <sip:3366 at 10.4.128.27>;tag=573234994
>>
>> Date: Fri, 22 Mar 2019 19:00:23 GMT
>>
>> Call-ID: 501227892-15 at 172.27.0.15
>>
>> CSeq: 1 INVITE
>>
>> Allow-Events: presence
>>
>> Warning: 304 10.4.128.27 "Media Type(s) Unavailable"
>>
>> Reason: Q.850;cause=65
>>
>> Content-Length: 0
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> SUCESSFULL CALL FROM CUCM
>>
>> -------------------------
>>
>> INVITE sip:*86329 at 172.27.0.12:5060
>> <https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2F86329%40172.27.0.12%3A5060&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628314437&sdata=olXlS7JOxjHDQbCleDv1CJq6yZi%2B8FEzIftvZ%2FIXu8A%3D&reserved=0>
>> SIP/2.0
>>
>> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
>>
>> From: "XXXX XXXX (3307)" <sip:3307 at 10.4.128.27
>> >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
>>
>> To: <sip:*86329 at 172.27.0.12>
>>
>> Date: Mon, 25 Mar 2019 10:40:36 GMT
>>
>> Call-ID: 6b366f80-c981b024-4f13-1b80040a at 10.4.128.27
>>
>> Supported: timer,resource-priority,replaces
>>
>> Min-SE:  1800
>>
>> User-Agent: Cisco-CUCM9.1
>>
>> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>> SUBSCRIBE, NOTIFY
>>
>> CSeq: 101 INVITE
>>
>> Expires: 180
>>
>> Allow-Events: presence, kpml
>>
>> Supported: X-cisco-srtp-fallback,X-cisco-original-called
>>
>> Cisco-Guid: 1798729600-0000065536-0000010811-0461374474
>>
>> Session-Expires:  1800
>>
>> P-Asserted-Identity: "XXXX XXXX (3307)" <sip:3307 at 10.4.128.27>
>>
>> Remote-Party-ID: "XXXX XXXX (3307)" <sip:3307 at 10.4.128.27
>> >;party=calling;screen=yes;privacy=off
>>
>> Contact: <sip:3307 at 10.4.128.27:5060
>> <https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fsip%3A3307%4010.4.128.27%3A5060&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628324445&sdata=RZBbBCLIvH5tdrdh3bGUzyVnNrGWExWeVHkcLfyFvAU%3D&reserved=0>
>> >
>>
>> Max-Forwards: 69
>>
>> Content-Type: application/sdp
>>
>> Content-Length: 212
>>
>> v=0
>>
>> o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27
>>
>> s=SIP Call
>>
>> c=IN IP4 10.4.128.12
>>
>> t=0 0
>>
>> m=audio 30530 RTP/AVP 8 101
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=ptime:20
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>>
>>
>>
>>
>> Answer from the PBX
>>
>> ----------------------
>>
>>
>>
>> SIP/2.0 183 Session Progress
>>
>> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8
>>
>> From: "Gabriel Querol (3307)" <sip:3307 at 10.4.128.27
>> >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
>>
>> To: <sip:*86329 at 172.27.0.12>;tag=43743456
>>
>> Call-ID: 6b366f80-c981b024-4f13-1b80040a at 10.4.128.27
>>
>> CSeq: 101 INVITE
>>
>> Server: MitE1x v4.4.5.1062
>>
>> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO
>>
>> P-Mitrol-idLlamada: 190325074112281_MIT_02447
>>
>> Content-Length: 217
>>
>> Content-Type: application/sdp
>>
>> v=0
>>
>> o=CiscoSystemsCCM-SIP 429005 1 IN IP4 172.27.0.12
>>
>> s=MitE1x Call
>>
>> c=IN IP4 172.27.0.12
>>
>> t=0 0
>>
>> m=audio 36508 RTP/AVP 8 101
>>
>> a=sendrecv
>>
>> a=rtpmap:8 PCMA/8000/1
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> a=fmtp:101 0-15
>>
>>
>>
>> _______________________________________________
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>>
>>
>>
>>
>> --
>>
>> For immediate assistance please reach out to Chemeketa IT Help Desk at
>> 5033997899
>>
>> -or-
>>
>> Visit the help center
>>
>>
>>
>> https://projects.chemeketa.edu/servicedesk/customer/portals
>> <https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fprojects.chemeketa.edu%2Fservicedesk%2Fcustomer%2Fportals&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628334454&sdata=V%2FCVqhnRGbXstML4SU4XxaKNA3SsfWpxii%2FyBedno8A%3D&reserved=0>
>>
>>
>>
>> Johnny Q
>>
>> Voice Technology Analyst II
>>
>> Network, Infrastructure, Routing Devices, and Servers
>>
>> Chemeketa Community College
>>
>> Johnny.Q at chemeketa.edu
>>
>> Building 22 Room 130
>>
>> Work 5033995294
>>
>> Mobile 9712182110
>>
>> SIP 5035406689
>>
>> FAX 5033995549
>>
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