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@ecn.net.au <support@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@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@X.Y.Z.A:5060 bridge( sofia/external/0737111111@A.B.C.X:5060) 2013-01-17 12:34:19.702102 [DEBUG] switch_channel.c:1089 sofia/aapt/0731111111@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@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@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@A.B.C.X:5060 [BREAK] 2013-01-17 12:34:19.702102 [DEBUG] switch_core_state_machine.c:415 ( sofia/external/0737111111@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@A.B.C.X:5060) State INIT 2013-01-17 12:34:19.702102 [DEBUG] mod_sofia.c:86 sofia/external/0737111111@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@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@A.B.C.D>;tag=mH93ZHryv4H2g To: <sip:0737111111@A.B.C.X:5060> Call-ID: 3ca1df3f-daf1-1230-f389-002219a7c712 CSeq: 38858965 INVITE Contact: <sip:mod_sofia@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@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@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@ecn.net.au <support@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@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@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely Jay