[cisco-voip] one way audio after SIP cupover

Thu Feb 8 12:44:14 EST 2018

The SIP Profile in CUCM does not apply to your CUBE, it applies to your
CUCM as it's speaking to the CUBE.  Don't forget to reset the trunk after
changing this, and be warned that all active calls would drop at that time.

Only configuration applied on the CLI of the CUBE will affect how CUBE
behaves.

Thanks to a forum post by Brian Meade
<https://supportforums.cisco.com/t5/ip-telephony/sip-udp-rtp-port-range/td-p/2423707>,
I found out, that the way you limit/change the RTP port range on your
gateway is:

voice service voip
 rtp-port range 20000 30000
!

However, that's global and not leg specific.

On Thu, Feb 8, 2018 at 9:02 AM Jonatan Quezada <
jonatan.quezada at chemeketa.edu> wrote:

> I was able to see these calls with these port open, but I thought the cube
> has its own internal set 16384 to 32766 for the inside facing my cucm, and
> the outside set range, per centuryLink 20000 to 59999.  how can I make sure
> that the cube is obeying the range in the SIP profile I have on the trunk
> that lives on the cube?
>
> [image: Inline image 1][image: Inline image 2]
>
> i feels like the matching is not happening, But also the CUCM should be
> matching in and out on its own or automagically.
> How can I verify that the sip profile and or trunk settings are configured
> to make sure this happens for each call?
>
>
> On Tue, Feb 6, 2018 at 12:19 PM, Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:
>
>> You can use this tutorial to see what steps are being processed in the
>> script, as well as what digits are being received if any.
>>
>>
>> https://supportforums.cisco.com/t5/collaboration-voice-and-video/uccx-viewing-executed-script-steps-via-cli/ta-p/3162231
>>
>> If you don't know what DTMF methods you're using on the outside and
>> inside of your network, then I would start by using the following command
>> on your CUBE and inspect the output during an active call:
>>
>> show call active voice | in PeerAddr|PeerId|RemoteS|RemoteM|Dtmf|Coder|VAD
>>
>>
>> On Tue, Feb 6, 2018 at 10:24 AM Jonatan Quezada <
>> jonatan.quezada at chemeketa.edu> wrote:
>>
>>> regarding the UCCX choking on calls, they are seeing more transfer drops
>>> than anything else. The other really bad part is, it seems that all of
>>> scripts are not registering the options when pressed it just drops the
>>> call. Is there more going on with the CTI ports, or rather an update to the
>>> configuration to handle the change in signalling from our provider?
>>>
>>> where else can I look to make sure that the UCCX is processing the calls
>>> from the sip trunk correctly?
>>>
>>> On Tue, Feb 6, 2018 at 6:57 AM, Ryan Huff <ryanhuff at outlook.com> wrote:
>>>
>>>> Good morning to you Mr. Holloway :)!
>>>>
>>>> DTMF: Generally speaking, mis-matched DTMF methods, in my experience,
>>>> presents with different symptoms than shown on that screenshot of the phone.
>>>>
>>>> SIP ACL: Correct, only applies to signaling, and if blocked, would
>>>> ultimately lead to a loss of RTP (but would then generally lead to the call
>>>> being disconnected and in most cases, stop the call from even setting up.
>>>> However, in a stressed CPU scenario (more common than you might think, ACL
>>>> application can be delayed). This would lead into my question about how
>>>> long the call was staying connected and what call flow that screen shot
>>>> depicts (in or out). I assume an inbound call based on the OP using the
>>>> word “caller”.
>>>>
>>>> The reason I suggest adding media and signaling addresses into the sip
>>>> ACL list is because some carriers will send signaling and media from the
>>>> same address or pool of addresses. Just did one not too long ago like that.
>>>>
>>>> Sent from my iPhone
>>>>
>>>> On Feb 6, 2018, at 9:22 AM, Anthony Holloway <
>>>> avholloway+cisco-voip at gmail.com> wrote:
>>>>
>>>> Ryan,
>>>>
>>>> For what you said here:
>>>>
>>>> *"Your call doesn’t appear to have a need for MTP or Transcoding (G711
>>>> both sides and matching sample sizes); so I wouldn’t start there."*
>>>>
>>>> Don't forget that DTMF relay needs to match too, and this is something,
>>>> in my opinion, that people miss-configure a lot!  In fact, I see people
>>>> with h245 alpha on their SIP dial peers?  Like what?  Typically, the SIP
>>>> ITSP will support RTP-NTE (RFC2833 [RFC 4733]) only, and your CUBE will
>>>> need to inter-work that DTMF with an OOB DTMF relay, such as SIP NOTIFY.
>>>> But then your SIP Trunk Sec Prof will need to allow Unsolicited
>>>> Notifications in order for that to work.  Also, some devices can support
>>>> RTP-NTE, but usually your CTI based apps cannot.  E.g., UCCX
>>>>
>>>> And for here:
>>>>
>>>> *"CUBE: ip trusted address list (make sure all provider signaling and
>>>> media addresses are authorized or ip authentication is off (which I do not
>>>> recommend) and make sure you include any CUCM addresses that are not used
>>>> in dial peers)."*
>>>>
>>>> Since this feature is just for signaling, and the call does establish,
>>>> this wouldn't be the cause of an RTP issue, and you wouldn't be putting
>>>> your media addresses in here.
>>>>
>>>> Do you agree with both of those remarks, or did I misunderstand
>>>> something?
>>>>
>>>> On Mon, Feb 5, 2018 at 5:14 PM Ryan Huff <ryanhuff at outlook.com> wrote:
>>>>
>>>>> Empirically, this “looks” like one way audio. How long will the call
>>>>> stay connected? Indefinitely? 30 seconds? 2 minutes?
>>>>>
>>>>> Your call doesn’t appear to have a need for MTP or Transcoding (G711
>>>>> both sides and matching sample sizes); so I wouldn’t start there.
>>>>>
>>>>> Check these items and see what you find;
>>>>>
>>>>> CUBE: ip trusted address list (make sure all provider signaling and
>>>>> media addresses are authorized or ip authentication is off (which I do not
>>>>> recommend) and make sure you include any CUCM addresses that are not used
>>>>> in dial peers).
>>>>>
>>>>> CUBE: double check your media and signal bindings and make sure they
>>>>> are binding correctly. Are you globally binding or dial peer binding?
>>>>>
>>>>> CUCM: verify the SIP trunk points to the CUBE interface that signaling
>>>>> is bound to (generally the same interface media would be bound to as well).
>>>>>
>>>>> CUBE:
>>>>> #logging buffered 10000000
>>>>> #enable debug ccsip messages
>>>>>
>>>>> Place a call and then look at the logs. Do you see any SIP error
>>>>> messages in the 4xx, 5xx (or more rare 6xx) range?
>>>>>
>>>>> As a quick gut check, if you can, enable “MTP Required” on the CUCM
>>>>> SIP trunk facing the CUBE (and make sure it has access to an MRGL/MRG that
>>>>> uses a CUCM node for MTP) and reset the trunk and test a call. If this
>>>>> works, it likely means you’re facing a network path issue between the
>>>>> phone’s IP network and the network of the CUBE interface facing CUCM.
>>>>>
>>>>> Outside of that, like Anthony said, it could be almost anything. A “sh
>>>>> run” or “sh tech” on the cube with a logging buffer from a ccsip messages
>>>>> during a failed call will generally get the ball rolling for most of us on
>>>>> this list in terms of offering targeted assistance.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Ryan
>>>>>
>>>>> On Feb 5, 2018, at 2:37 PM, Anthony Holloway <
>>>>> avholloway+cisco-voip at gmail.com> wrote:
>>>>>
>>>>> The fact that you received 2 packets is interesting.  Tells me that
>>>>> there is routing happening correctly...to some degree.
>>>>>
>>>>> If you go to the web page of the phone and click on stream 1, does the
>>>>> far end IP address match your CUBE address?
>>>>>
>>>>> Also, there's a lot of settings that need to be considered when
>>>>> implementing SIP, such as:
>>>>>
>>>>> Early Offer and MTP usage
>>>>> PRACK/Early Media
>>>>> Offfer/Answer (Capabilities)
>>>>> Interface Binding
>>>>> Transport Protocol
>>>>> OPTIONS Ping
>>>>> Duplex Streaming
>>>>> Midcall Signaling
>>>>> Timers
>>>>> etc.
>>>>>
>>>>> Depending on your setting, a lot of different possibilities exist for
>>>>> why you might have the experience you have.  If you could paint a clearer
>>>>> picture of your scenario, that might help out.
>>>>>
>>>>> On Fri, Feb 2, 2018 at 5:47 PM Jonatan Quezada <
>>>>> jonatan.quezada at chemeketa.edu> wrote:
>>>>>
>>>>> I get that this is usually routing but, is it also routing when the
>>>>>> issue is intermittent?
>>>>>>
>>>>>> our call flow is like so
>>>>>>
>>>>>> CentLink(Provider)
>>>>>> ----siptrunk30Meg-PPP(IQ-private)---Cube---CUCM10.5, uccx,unity
>>>>>>
>>>>>> <image.png>
>>>>>>
>>>>>> bonus facts, I have an operator who is in one of the two most
>>>>>> affected buildings and she can recover the call after hold,
>>>>>> resume,hold,resume sequence. then full rtp stream is there and she can hear
>>>>>> and speak with caller.
>>>>>>
>>>>>> are there SIP state change timers I can adjust, I want to tread
>>>>>> lightly though because out of all of our outreachs seperated by a metro
>>>>>> ethernet hub and spoke topology and almost 30 buildings here on main campus
>>>>>> only 2 seem to be affected.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> For immediate assistance please reach out to Chemeketa IT Help Desk
>>>>>> at 5033997899 <(503)%20399-7899>
>>>>>> -or-
>>>>>> Visit the help center from your employee dashboard found here:
>>>>>> *https://dashboard.chemeketa.edu/helpcenter/default.aspx
>>>>>> <https://dashboard.chemeketa.edu/helpcenter/default.aspx>*
>>>>>>
>>>>>>
>>>>>> Johnny Q
>>>>>> Voice Technology Analyst - TelNet
>>>>>> Chemeketa Community College
>>>>>> Johnny.Q at chemeketa.edu
>>>>>> Building 22 Room 131
>>>>>> Work 5033995294 <(503)%20399-5294>
>>>>>> Mobile 9712182110 <(971)%20218-2110>
>>>>>> SIP 5035406686 <(503)%20540-6686>
>>>>>>
>>>>> _______________________________________________
>>>>>> cisco-voip mailing list
>>>>>> cisco-voip at puck.nether.net
>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>
>>>
>>> --
>>> For immediate assistance please reach out to Chemeketa IT Help Desk at
>>> 5033997899 <(503)%20399-7899>
>>> -or-
>>> Visit the help center from your employee dashboard found here:
>>> *https://dashboard.chemeketa.edu/helpcenter/default.aspx
>>> <https://dashboard.chemeketa.edu/helpcenter/default.aspx>*
>>>
>>>
>>> Johnny Q
>>> Voice Technology Analyst - TelNet
>>> Chemeketa Community College
>>> Johnny.Q at chemeketa.edu
>>> Building 22 Room 131
>>> Work 5033995294 <(503)%20399-5294>
>>> Mobile 9712182110 <(971)%20218-2110>
>>> SIP 5035406686 <(503)%20540-6686>
>>>
>>
>
>
> --
> For immediate assistance please reach out to Chemeketa IT Help Desk at
> 5033997899 <(503)%20399-7899>
> -or-
> Visit the help center from your employee dashboard found here:
> *https://dashboard.chemeketa.edu/helpcenter/default.aspx
> <https://dashboard.chemeketa.edu/helpcenter/default.aspx>*
>
>
> Johnny Q
> Voice Technology Analyst - TelNet
> Chemeketa Community College
> Johnny.Q at chemeketa.edu
> Building 22 Room 131
> Work 5033995294 <(503)%20399-5294>
> Mobile 9712182110 <(971)%20218-2110>
> SIP 5035406686 <(503)%20540-6686>
>
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