Study this example:
http://sipp.sourceforge.net/doc/uac_pcap.xml.html
I think it has everything you need.
It sends out of band DTMF digit using G711 a-law and you can adapt to your
case
Don't go for DTMF in INFO unless this is what you want to test. DTMF in INFO
is not a very common practice, prefer out of band DTMF as RC2833.....
On Fri, Aug 19, 2011 at 11:29 AM, Corcione, John
<[email protected]>wrote:
> Mayama,****
>
> ** **
>
> This is my invite to the IVR Platform – How is SIP INFO used in this case.
> And how do I force the IVR to accept the codec? Should I also add a line to
> the a=rtpmap: for G711 as well?****
>
> ** **
>
> INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0****
>
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]****
>
> From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]****
>
> To: <sip:[service]@[remote_ip]:[remote_port]>****
>
> Call-ID: [call_id]****
>
> CSeq: 1 INVITE****
>
> Supported: 100rel, replaces****
>
> Max-Forwards: 70****
>
> Accept: application/sdp****
>
> Allow: INVITE, ACK, CANCEL, OPTIONS,BYE, UPDATE, REFER, NOTIFY****
>
> Contact: <sip:sipp@[local_ip]:[local_port];transport=[transport]>****
>
> Content-Type: application/sdp****
>
> Content-Length: [len]****
>
> ** **
>
> v=0****
>
> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]****
>
> s=SIP Media Capabilities****
>
> c=IN IP[media_ip_type] [media_ip]****
>
> t=0 0****
>
> m=audio [media_port] RTP/AVP 18 0 2 96****
>
> a=rtpmap:18 G729/8000****
>
> a=rtpmap:2 G726-32/8000****
>
> a=rtpmap96 telephone-event/8000****
>
> a=fmtp:96 0-15****
>
> a=fmtp:18 annexb=no****
>
> a=fmtp:96 0-15****
>
> a=maxptime:20****
>
> a=sendrecv]]>****
>
> </send>****
>
> <recv response="100" crlf="true" />****
>
> <recv response="183" crlf="true" />****
>
> <send>****
>
> ** **
>
> *From:* mayamatakeshi [mailto:[email protected]]
> *Sent:* Thursday, August 18, 2011 8:14 PM
> *To:* Corcione, John
> *Cc:* [email protected]
> *Subject:* Re: [Sipp-users] Entering DTMF in the SIPP Scenario PLEASE READ
> ****
>
> ** **
>
> ** **
>
> On Fri, Aug 19, 2011 at 5:13 AM, Corcione, John <[email protected]>
> wrote:****
>
> I am curious to know.
>
> XML file plays a dtmf CC# and PIN, however when my IVR telecom person
> checks the IVR application he advises that the Voice Portal Application log
> says it is not receiving any digits however the messaging in the scenario
> seems to be working invite 100, 183, prack , 200, 200 ack, bye etc.... Can
> anyone explain this?****
>
>
> Nothing to be surprised about: SIPp deals with SIP messages so it can only
> check if SIP transactions were successful. DTMF can be transmitted using RTP
> (in-band, out-of-band) or using SIP INFO. So assuming you are not using SIP
> INFO for this (as you have not mentioned it), then you are using RTP and so
> SIPp have no way to tell if the digits were received or not.
> SIPp just offers the feature of playing a pcap file containing whatever you
> want but you are on your own to check if they are being properly processed
> by the distant-end.
>
> To identify the problem, you need to inspect a packet capture from the IVR
> side.
> I can imagine the following possibilities:
> - packets are not reaching the IVR (blocked UDP ports)
> - if you are using in-band DTMF, you might have negotiated one codec, but
> the pcap file contains packets for a different codec (for example, if you
> offer more than one codec in your INVITE, you must check the SDP from the
> IVR and play a different file depending on which codec the IVR selected).
> (Actually, in this case, I recommend to send just one codec in the INVITE so
> that the distant-end will have no choice).
> - if you are using out-of-band DTMF, the payload-type for telephone-event
> is mismatched (for example, you offered it as 101 and the packets in the
> pcap file are set as such, but the IVR answered setting it to another value
> like 112 and so it is discarding your packets as garbage).
> - codec negotiation is fine but you are using a codec which cannot
> transmit DTMF reliably.
>
> regards,
> takeshi****
>
>
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