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

Tue Mar 26 12:39:53 EDT 2019

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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>
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>
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>
> Johnny Q
>
> Voice Technology Analyst II
>
> Network, Infrastructure, Routing Devices, and Servers
>
> Chemeketa Community College
>
> Johnny.Q at chemeketa.edu
>
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>
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>
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>
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