[cisco-voip] Jabber region restriction issue

Anthony Holloway avholloway+cisco-voip at gmail.com
Mon Oct 8 00:16:56 EDT 2018


Bernhard, Good job proposing an MTP is being invoked, and I would say the
same.  There's a number of places/reasons an MTP could be inserted, how
would you systematically check this?  I.e., What's your approach?

Also, we don't have to assume .2 is a CUCM node.  Look at the SIP Via:
header.  The call flow was was described as: Jabber > CUCM > CUBE > SP, and
the SIP request is label as being "CUBE received."

Not too mention the handful of other lines in that message which all point
to .2 being a CUCM node. (SDP o line, SIP User-Agent header, etc.)



On Sun, Oct 7, 2018 at 10:22 PM Bernhard Albler <bernhard.albler at gmail.com>
wrote:

> looking at your invite it looks like an MTP is being invoked ( assuming .2
> is the address of a cucm node)
> Thats the reason g711 is being used
> now the question is if the MTP was inserted via config or because of some
> ( e.g. dtmf) other reason and if it is safe to remove it therefore...
>
> On Mon 8. Oct 2018 at 00:21, Carlos G Mendioroz via cisco-voip <
> cisco-voip at puck.nether.net> wrote:
>
>> Hi,
>> kind of rusty (me, long time since engaging in troubleshooting of Voip)
>> but I encountered something weird (again, may be me).
>>
>> Jabber Android (12.1) registered to CUCM (10.5) with region set with 16K
>> audio limit.
>>
>> Call comes from Jabber through CUCM to CUBE to SP, but signalled as G711
>> and after session progress, it gets cancelled.
>>
>> Shouldn't CUCM signal the call with G.729 in the first place ?
>>
>> CUBE received:
>> INVITE sip:947930020 at 10.0.100.1:5060 SIP/2.0
>> Via: SIP/2.0/TCP 10.0.100.2:5060;branch=z9hG4bK235a8e855ad
>> From:
>> <sip:3560 at 10.0.100.2
>> >;tag=36658~4fdda745-cceb-4407-a170-1420de65d7d7-22052972
>> To: <sip:947930020 at 10.0.100.1>
>> Date: Sun, 07 Oct 2018 20:45:59 GMT
>> Call-ID: fd98a300-bba17087-1b99-264000a at 10.0.100.2
>> Supported: timer,resource-priority,replaces
>> Min-SE:  1800
>> User-Agent: Cisco-CUCM10.5
>> 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: 4254638848-0000065536-0000000690-0040108042
>> Session-Expires:  1800
>> P-Asserted-Identity: <sip:3560 at 10.0.100.2>
>> Remote-Party-ID: <sip:3560 at 10.0.100.2
>> >;party=calling;screen=yes;privacy=off
>> Contact:
>> <sip:3560 at 10.0.100.2:5060;transport=tcp>;+u.sip!devicename.ccm.cisco.com
>> ="BOTTRON"
>> Max-Forwards: 13
>> Content-Type: application/sdp
>> Content-Length: 197
>>
>> v=0
>> o=CiscoSystemsCCM-SIP 36658 1 IN IP4 10.0.100.2
>> s=SIP Call
>> c=IN IP4 10.0.100.2
>> t=0 0
>> m=audio 26804 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>>
>>
>> TIA,
>> --
>> Carlos G Mendioroz  <tron at huapi.ba.ar>  LW7 EQI  Argentina
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> --
> Bernhard Albler, +4369917207384
> --
> "Was Nachwelt! Wie komm' ich dazu was für die Nachwelt zu tun? Was hat
> denn die Nachwelt für mich getan?"
> --Carl Friedrich Zelter
> _______________________________________________
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>
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