[cisco-voip] Jabber region restriction issue
Kent Roberts
kent at fredf.org
Mon Oct 8 12:17:26 EDT 2018
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
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