[cisco-voip] Jabber region restriction issue

Carlos G Mendioroz tron at huapi.ba.ar
Mon Oct 8 14:21:05 EDT 2018


FTR, this proved to be tough on me.
And I thought I knew something about it :)

Ended up removing a force rtp-nte that was there, and adding sip-kpml as
an alternative on the CUBE (performing as a CME in this case too).

And changed the sip profile on the CUCM to do Early offer (best effort).

HTH someone in the future.
-Carlos

Kent Roberts @ 08/10/2018 13:17 -0300 dixit:
> You might update your dial-peers to use G729 and G711 only.     Not all carriers support G722.   Or put it as 3rd of 4th option.    Also might try early offer on the cube.
> 
> 
> 
> 
>> On Oct 8, 2018, at 10:10 AM, Carlos G Mendioroz via cisco-voip <cisco-voip at puck.nether.net> wrote:
>>
>> Grr,
>> now that MTP is not forced, calls from CUCM phones to some dialpeers
>> (phones) on the CUBE fail :(
>>
>> The only weirdness seems to be:
>>
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 2983 2791 IN IP4 10.0.100.1
>> s=SIP Call
>> c=IN IP4 10.0.100.1
>> t=0 0
>> m=audio 18746 RTP/AVP 9 116 0 8 18 101
>> c=IN IP4 10.0.100.1
>> a=rtpmap:9 G722/8000
>> a=fmtp:9 bitrate=64
>> a=rtpmap:116 iLBC/8000
>> a=fmtp:116 mode=20
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>
>> versus:
>>
>> v=0
>> o=CiscoSystemsCCM-SIP 38001 1 IN IP4 10.0.100.2
>> s=SIP Call
>> c=IN IP4 10.0.100.5
>> b=TIAS:64000
>> b=AS:64
>> t=0 0
>> m=audio 49328 RTP/AVP 9 101
>> a=rtpmap:9 G722/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=rtcp:49329 IN IP4 10.0.100.5
>>
>> (note 101 0-16 and 0-15 missmatch).
>>
>> The CUBE is tearing down with:
>>
>> Reason: Q.850;cause=172
>>
>>
>>
>> Carlos G Mendioroz via cisco-voip @ 08/10/2018 07:25 -0300 dixit:
>>> Indeed, an MTP was forced by the SIP trunk config.
>>> I'll have to rethink my MTP strategy, because having had so many issues
>>> with NOT having an MTP, I'm used to insert MTPs by default on SIP trunks.
>>>
>>> Still not comfortable though, CUCM knows it's going to a dead end and
>>> waits untill progress to "bail out". But I can understand why now (there
>>> could have been xcodes to make the job, there are none).
>>>
>>> Thanks!!!
>>>
>>> Anthony Holloway @ 08/10/2018 01:16 -0300 dixit:
>>>> 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 <mailto: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 <mailto: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
>>>>        <http://sip:947930020@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
>>>>        <mailto:sip%3A3560 at 10.0.100.2>>;tag=36658~4fdda745-cceb-4407-a170-1420de65d7d7-22052972
>>>>        To: <sip:947930020 at 10.0.100.1 <mailto:sip%3A947930020 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
>>>>        <mailto: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
>>>>        <mailto:sip%3A3560 at 10.0.100.2>>
>>>>        Remote-Party-ID: <sip:3560 at 10.0.100.2
>>>>        <mailto:sip%3A3560 at 10.0.100.2>>;party=calling;screen=yes;privacy=off
>>>>        Contact:
>>>>        <sip:3560 at 10.0.100.2:5060
>>>>        <http://sip:3560@10.0.100.2:5060>;transport=tcp>;+u.sip!devicename.ccm.cisco.com
>>>>        <http://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
>>>>        <mailto:tron at huapi.ba.ar>>  LW7 EQI  Argentina
>>>>        _______________________________________________
>>>>        cisco-voip mailing list
>>>>        cisco-voip at puck.nether.net <mailto: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
>>>>    _______________________________________________
>>>>    cisco-voip mailing list
>>>>    cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>>>>    https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>
>> -- 
>> 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
> 

-- 
Carlos G Mendioroz  <tron at huapi.ba.ar>  LW7 EQI  Argentina


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