[cisco-voip] DTMF SIP to Verizon, wrong payload type...
Anthony Holloway
avholloway+cisco-voip at gmail.com
Thu May 17 14:30:45 EDT 2012
I'm pretty sure he's wrong.
>From the RFC:
The connection (`c=') and attribute (`a=') information in the
> session-level section applies to all the media of that session unless
> overridden by connection information or an attribute of the same name
> in the media description.
>
> The media
> description starts with an `m=' line and continues to the next media
> description or end of the whole session description. In general,
> session-level values are the default for all media unless overridden
> by an equivalent media-level value.
>
>
http://www.ietf.org/rfc/rfc2327.txt
And here's a link to the Media Description definition:
http://tools.ietf.org/html/rfc4566#page-22
Here's a Stackoverflow discussion on the m= field, and it confirms that the
trailing 101 is the telehony event for DTMF:
http://stackoverflow.com/questions/2930288/sdp-media-field-format
Since you are a partner, I will look for a recent VoE session which also
talk about the same thing...looking now.
-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
>>>>>
>>>>>
>>>>
>>>
>>
>
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