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

Mon Apr 1 17:04:01 EDT 2019

In the end, my customer finally realized the problema was on the PBX side. The unupgraded PBX worked fine.

Thanks Brian for your help.

Regards,

Ariel.

De: cisco-voip <cisco-voip-bounces at puck.nether.net> En nombre de ROZA, Ariel
Enviado el: martes, 26 de marzo de 2019 14:04
Para: Brian Meade <bmeade90 at vt.edu>
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

´ll check with my customer, and report back.
I saw that negative parameter on the o= line, but I wasn´t completely certain how to handle it.
Thanks for the help!

De: Brian Meade <bmeade90 at vt.edu<mailto:bmeade90 at vt.edu>>
Enviado el: martes, 26 de marzo de 2019 13:56
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM<mailto:Ariel.ROZA at LA.LOGICALIS.COM>>
CC: cisco-voip (cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>) <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade

Actually meant o= line is the origin line.

On Tue, Mar 26, 2019 at 12:39 PM Brian Meade <bmeade90 at vt.edu<mailto: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<mailto:sip%3A3366 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<mailto:sip%3A86329 at 172.27.0.15>>;tag=2792862
To: <sip:3366 at 10.4.128.27<mailto:sip%3A3366 at 10.4.128.27>>
Call-ID: 501227892-15 at 172.27.0.15<mailto: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<mailto:sip%3A86329 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<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Frfc4566%23section-5.2&data=02%7C01%7Cariel.roza%40la.logicalis.com%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878451827&sdata=YILRc5UxLiA30sXHmbaGaM9AAyJcHE7P4mc2VEjM8To%3D&reserved=0>

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<mailto:Ariel.ROZA at la.logicalis.com>> wrote:
Here´s the trace file with the bad call

De: Brian Meade <bmeade90 at vt.edu<mailto:bmeade90 at vt.edu>>
Enviado el: lunes, 25 de marzo de 2019 23:39
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM<mailto:Ariel.ROZA at LA.LOGICALIS.COM>>
CC: Jonatan Quezada <jonatan.quezada at chemeketa.edu<mailto:jonatan.quezada at chemeketa.edu>>; cisco-voip (cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>) <cisco-voip at puck.nether.net<mailto: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<mailto: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<mailto:jonatan.quezada at chemeketa.edu>>
Enviado el: lunes, 25 de marzo de 2019 19:24
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM<mailto:Ariel.ROZA at LA.LOGICALIS.COM>>
CC: cisco-voip (cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>) <cisco-voip at puck.nether.net<mailto: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<mailto: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<mailto:sip%3A3366 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<mailto:sip%3A86329 at 172.27.0.15>>;tag=2792862
To: <sip:3366 at 10.4.128.27<mailto:sip%3A3366 at 10.4.128.27>>
Call-ID: 501227892-15 at 172.27.0.15<mailto: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<mailto:sip%3A86329 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<mailto:sip%3A86329 at 172.27.0.15>>;tag=2792862
To: <sip:3366 at 10.4.128.27<mailto:sip%3A3366 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<mailto: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%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878461836&sdata=NveSUI5N%2FNPWtQ5j7yCw7a%2BcGyv%2Bvpf6tSoJ3ywRKBw%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<mailto:sip%3A3307 at 10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: <sip:*86329 at 172.27.0.12<mailto: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<mailto: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<mailto:sip%3A3307 at 10.4.128.27>>
Remote-Party-ID: "XXXX XXXX (3307)" <sip:3307 at 10.4.128.27<mailto:sip%3A3307 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%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878461836&sdata=J9V6cLg3fBgJsrLxbc1%2FZJTL3%2BQm2899EgwSFmS7jbY%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<mailto:sip%3A3307 at 10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: <sip:*86329 at 172.27.0.12<mailto:86329 at 172.27.0.12>>;tag=43743456
Call-ID: 6b366f80-c981b024-4f13-1b80040a at 10.4.128.27<mailto: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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