[cisco-voip] DTMF SIP to Verizon, wrong payload type...

[Anthony Holloway]

Here is the VoE I was talking about:

https://communities.cisco.com/docs/DOC-7823

Look towards the top for "VoE - CUBE SIP Trunking"

Then download the PDF, and goto page 90.  The page is also discuss in the
Webex recording @ 1h 48m 55s.  For those who cannot see this, it says:

“c” parameter identifies the IP
> address (20.1.1.1) that the peer
> device should send the media to
>


> “m” parameter identifies:
> the type of call (audio)
> port number for media (16950)
> payload type for the 1st
> preferred codec (18 for G729)
> dtmf (101 for RFC2833)
>


> “a’” parameter identifies all the
> codecs and other descriptors for this
> call leg


This VoE event is very informative.  Hope that helps.

-Anthony

On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip at gmail.com> wrote:

> It is not.
>
> Per Verizon tech:
>
>      Octet1058 SIP Message Body: SDP
>
> --------------------------------------------------------------------------------
>      ........  Header Field           v=0
>      ........                         o=CiscoSystemsSIP-GW-UserAgent 794
> 632 IN IP4 1,1,1,1
>      ........                         s=SIP Call
>      ........                         c=IN IP4 1.1.1.1
>      ........                         t=0 0
>      ........                         m=audio 17176 RTP/AVP 0 101
>      ........                         c=IN IP4 1.1.1.1
>      ........                         a=rtpmap:0 PCMU/8000
>      ........                         a=rtpmap:101 X-NSE/8000   <-- should
> be telephone-event/8000
>      ........                         a=fmtp:101 192-194
>      ........                         a=ptime:20
>
> They say the problem is on our end, and since we are sending the wrong
> DTMF, they are closing their ticket.
>
>
>
>
> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:
>
>> I'm glad you posted that.
>>
>> The m= is the actual setting for that call.  The a= are the available
>> settings.  And you can see in the m=, you have codec 0 (g711) and DTMF 101
>> (telephony).
>>
>>  This looks correct.
>>
>> -Anthony
>>
>>
>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip at gmail.com>wrote:
>>
>>> Added it, no change.
>>>
>>>
>>> v=0
>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>> s=SIP Call
>>> c=IN IP4 1.1.1.1
>>> t=0 0
>>> m=audio 18130 RTP/AVP 0 101
>>> c=IN IP4 157.130.97.178
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 X-NSE/8000               <------------- this needs to be
>>> telephone-event/8000
>>> a=fmtp:101 192-194
>>> a=ptime:20
>>>
>>>
>>>
>>>
>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>> avholloway+cisco-voip at gmail.com> wrote:
>>>
>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>  What is your DTMF Signaling Method set to on that Trunk?
>>>>
>>>> The only command I run which I can see is missing from your config is:
>>>>
>>>> voice service voip
>>>>   dtmf-interworking rtp-nte
>>>>
>>>> But I'm not positive that's your problem.
>>>>
>>>> -Anthony
>>>>
>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip at gmail.com>wrote:
>>>>
>>>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>>>> work fine, however, for toll free calls, DTMF does not function.
>>>>>
>>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>>>> this:
>>>>>
>>>>> a=rtpmap:101 X-NSE/8000
>>>>>
>>>>> And it should be:
>>>>>
>>>>> telephone-event/8000
>>>>>
>>>>> And that is why it is failing.
>>>>>
>>>>>
>>>>>
>>>>> What configuration change can we do to force it to send the right DTMF
>>>>> method?
>>>>>
>>>>>
>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>
>>>>> Relevant SIP Config:
>>>>>
>>>>>
>>>>> !
>>>>> voice call send-alert
>>>>> voice rtp send-recv
>>>>> !
>>>>> voice service voip
>>>>>  allow-connections h323 to h323
>>>>>  allow-connections h323 to sip
>>>>>  allow-connections sip to h323
>>>>>  allow-connections sip to sip
>>>>>  no supplementary-service sip refer
>>>>>  redirect ip2ip
>>>>>  h323
>>>>>   h225 display-ie ccm-compatible
>>>>>  modem passthrough nse payload-type 101 codec g711ulaw
>>>>>  sip
>>>>>   bind media source-interface MFR1
>>>>>   early-offer forced
>>>>>   midcall-signaling passthru
>>>>> !
>>>>> !
>>>>>
>>>>> dial-peer voice 800 voip
>>>>>  description OUTBOUND Voice SIP calls to VzB
>>>>>  destination-pattern 1800[2-9]......
>>>>>  voice-class sip dtmf-relay force rtp-nte
>>>>>  session protocol sipv2
>>>>>  session target sip-server
>>>>>  incoming called-number .
>>>>>  dtmf-relay rtp-nte
>>>>>  codec g711ulaw
>>>>>  no vad
>>>>>
>>>>>
>>>>> !
>>>>> sip-ua
>>>>>  retry invite 2
>>>>>  retry bye 2
>>>>>  retry cancel 2
>>>>>  registrar dns:verizonsipgateway expires 3600
>>>>>  sip-server dns:verizonsipgateway:5071
>>>>>  g729-annexb override
>>>>> !
>>>>>
>>>>>
>>>>>  Jonathan
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20120517/7ddd3da4/attachment.html>