[Freeswitch-users] SBC In-band DTMF
jay binks
jaybinks at gmail.com
Thu Jan 17 07:16:50 MSK 2013
the problem with doing the sip traces how you have, is that you dont have
the RTP Stream.
id be using tcpdump on your boxes, so you can see does the RTP ( that
appears to be cut off ) even get to your FS node.
http://wiki.freeswitch.org/wiki/Packet_Capture
do something like
tcpdump -nq -s 0 -i eth0 -w /tmp/dump.pcap
On 17 January 2013 12:58, support at ecn.net.au <support at ecn.net.au> wrote:
> Hi
>
> This sounds quite posible.
>
> I've tcpdumped sip headers on both the freeswitch and the backend PBX.
>
> The only significant difference I can see between our old SBC and FS is
> that the old SBC
> asserted a media attribute fmtp:18 annexb=no where as FS doesn't forward
> this.
>
> To test it out we have set dtmf_mode to none on both contexts (the context
> facing the telco
> and the context facing the pbx), and set
>
> <param name="rfc2833-pt" value="0"/>
>
> (this may not be needed).
>
> then in the dialplan we have
>
> <action application="export" data="sip_append_audio_sdp=a=fmtp:18
> annexb=no"/>
>
> set prior to the bridge.
>
> --
>
> When we sip trace however we are not appending the attribute to the SDP on
> the INVITE.
>
> Logs and sip headers below:
>
>
> EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060export(sip_append_audio_sdp=a=fmtp:18 annexb=no)
> 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1135 EXPORT
> (export_vars) [sip_append_audio_sdp]=[a=fmtp:18 annexb=no]
> EXECUTE sofia/aapt/0731111111 at X.Y.Z.A:5060 bridge(
> sofia/external/0737111111 at A.B.C.X:5060)
> 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089
> sofia/aapt/0731111111 at X.Y.Z.A:5060 EXPORTING[export_vars]
> [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to eve
> nt
> 2013-01-17 12:34:19.702102 [DEBUG] switch_ivr_originate.c:2022 Parsing
> global variables
> 2013-01-17 12:34:19.702102 [NOTICE] switch_channel.c:968 New Channel
> sofia/external/0737111111 at A.B.C.X:5060[a30c650a-87d3-4027-9010-213547d698aa]
> 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:4977 (
> sofia/external/0737111111 at A.B.C.X:5060) State Change CS_NEW -> CS_INIT
> 2013-01-17 12:34:19.702102 [DEBUG] switch_core_session.c:1283 Send signal
> sofia/external/0737111111 at A.B.C.X:5060 [BREAK]
> 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 (
> sofia/external/0737111111 at A.B.C.X:5060) Running State Change CS_INIT
> 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:454 (
> sofia/external/0737111111 at A.B.C.X:5060) State INIT
> 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86
> sofia/external/0737111111 at A.B.C.X:5060 SOFIA INIT
> 2013-01-17 12:34:19.702102 [DEBUG] sofia_glue.c:2647 Local SDP:
> v=0
> o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D
> s=FreeSWITCH
> c=IN IP4 A.B.C.D
> t=0 0
> m=audio 24552 RTP/AVP 8 0 3 13
> a=fmtp:18 annexb=no
> a=ptime:20
> a=sendrecv
>
>
> But then the actual Invite doesn't contain the attribute in the SDP
>
> send 1036 bytes to udp/[A.B.C.X]:5060 at 02:34:19.718385:
> ------------------------------------------------------------------------
> INVITE sip:0737111111 at A.B.C.X:5060 SIP/2.0
> Via: SIP/2.0/UDP A.B.C.D:5080;rport;branch=z9hG4bKpUvH6K1ppUSNp
> Max-Forwards: 69
> From: "0731111111" <sip:0731111111 at A.B.C.D>;tag=mH93ZHryv4H2g
> To: <sip:0737111111 at A.B.C.X:5060>
> Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712
> CSeq: 38858965 INVITE
> Contact: <sip:mod_sofia at A.B.C.D:5080>
> User-Agent: FreeSWITCH-mod_sofia/1.2.5.3
> +git~20121229T001759Z~e04eab7902
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 155
> X-Nortel-Profile: DEFAULT
> X-FS-Support: update_display,send_info
> Remote-Party-ID: "0731111111" <sip:0731111111 at A.B.C.D
> >;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1358365507 1358365508 IN IP4 A.B.C.D
> s=FreeSWITCH
> c=IN IP4 A.B.C.D
> t=0 0
> m=audio 24552 RTP/AVP 8 0 3 13
> a=ptime:20
> ------------------------------------------------------------------------
>
>
>
> Are we missing something do you think?
>
>
>
> Kind Regards,
>
> ------------------------------
> *From:* Steven Ayre [steveayre at gmail.com]
> *Sent:* Thursday, 17 January 2013 4:14 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SBC In-band DTMF
>
> However under freeswitch if we don't start_dtmf before the bridge the
>> backend PBX boxes don't recognise
>> the DTMF inband (even though the tones are audible ie you can hear them
>> on a call recording on the
>> PBX).
>>
>> Have we missed something here? We would have thought with inband DTMF on
>> non compressed codec (no
>> transcoding) that the tones would just work with the media stream?
>
>
> start_dtmf will detect inband DTMF, and then send out of band on the
> outgoing leg of the bridge. This is expected. You'll will need to call
> start_dtmf from dialplan, which can easily be done from dialplan by
> checking for some condition identifying calls requiring it. If they're
> authenticating to FS then it'd be trivial to set a variable in their user
> directory entry that you can test in the dialplan.
>
> Now this is just a guess... During the codec negotiation FS will tell
> your PBX that it supports telephone-event (RFC2833). It may be that because
> the PBX sees that in the SDP that it doesn't look for inband DTMF, while
> when it received the SDP direct from the customer it didn't contain
> telephone-event and so did check for inband DTMF. Indeed it'd probably be a
> good idea for them to do so, since if you receive the same DTMF digit both
> inband and through RFC2833 then you'd be duplicating digits. So my guess
> is that the PBX doesn't bother check for inband because FS tells it it's
> sending out-of-band, which'd differ from before.
>
> -Steve
>
>
>
> -Steve
>
>
>
>
> On 16 January 2013 04:02, support at ecn.net.au <support at ecn.net.au> wrote:
> > Hi All
> >
> >
> >
> > We're quite new to Freeswitch and are in the process of migrating from
> > OpenSer (as an SBC) to Freeswitch.
> >
> >
> >
> > Mostly all is working well, except an oddity on DTMF.
> >
> >
> >
> > Our scenario:
> >
> >
> >
> > Telco/SIP Provider A is passing us calls using DTMF inband.
> >
> >
> >
> > We have a freeswitch configured as a SBC using 2 sip profiles (telco and
> > internal) to topology hide and manage
> >
> > distribution of calls to the PBX servers located behind the SBC.
> >
> >
> >
> > The freeswitch will be handling up to a few hundred calls so we're
> trying to
> > keep it lightweight.
> >
> >
> >
> > Behind the SBC is a series of Asterisk and Freeswitch PBX boxes handling
> > customer needs.
> >
> >
> >
> > An example inbound call profile looks like this:
> >
> >
> >
> > <extension name="Inbound 124356">
> >
> > <condition field="destination_number" expression="^(123456)$">
> >
> > <action application="pre_answer"/>
> >
> > <action application="start_dtmf" />
> >
> > <action application="bridge"
> > data="sofia/external/123456 at INTERNAL.PBX.IP:5060"/>
> >
> > </condition>
> >
> > </extension>
> >
> >
> >
> > Initially when calling into the platform IVR type applications runinng on
> > our PBX boxes would not
> >
> > work (you could hear the DTMF but the platform did not recognise the
> tones).
> >
> >
> >
> > We have had to add the appliation start_dtmf in order for Freeswitch to
> pass
> > the DTMF to the Asterisk
> >
> > PBX behind the SBC. Interestingly on our OpenSer platform we just
> proxied
> > the media (rtpproxy) with
> >
> > inband DTMF from the Telco and our PBX boxes recognised the inband DTMF
> > tones on the PBX platforms and
> >
> > IVR type applications just worked.
> >
> >
> >
> > However under freeswitch if we don't start_dtmf before the bridge the
> > backend PBX boxes don't recognise
> >
> > the DTMF inband (even though the tones are audible ie you can hear them
> on a
> > call recording on the
> >
> > PBX).
> >
> >
> >
> > Have we missed something here? We would have thought with inband DTMF on
> > non compressed codec (no
> >
> > transcoding) that the tones would just work with the media stream?
> >
> >
> >
> > We have confirmed both legs are PCMA and when using start_dtmf the first
> > second of the call is clipped.
> >
> >
> >
> >
> >
> >
> >
> > Kind Regards,
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> >
> >
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
--
Sincerely
Jay
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