[Freeswitch-users] one-way audio after playback+bridge
Artem Shiyanov
shiyanov at gmail.com
Fri Jun 26 10:59:38 PDT 2009
Updates:
1. One-way audio is in 95% tries. But how the rest 5% works??
2. Strange FS logging after the channels are bridged (user A talk to user B)
2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1005 at 192.168.147.1 entering state [ready]
2009-06-26 02:16:07 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4 192.168.147.130
s=FreeSWITCH
c=IN IP4 192.168.147.130
t=0 0
m=audio 31134 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
m=video 0 RTP/AVP 34
a=rtpmap:34 H263/90000
2009-06-26 02:16:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1000000000 at 192.168.147.130:5060 entering state [ready]
freeswitch at localhost.localdomain> 2009-06-26 02:17:09 [DEBUG] sofia.c:2728
sofia_handle_sip_i_state() Channel sofia/external/
1005 at uat.pbx.starpoundtech.net entering state [calling]
2009-06-26 02:17:09 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready]
2009-06-26 02:17:09 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=- 1 3 IN IP4 192.168.147.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.147.1
t=0 0
m=audio 47590 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
freeswitch at localhost.localdomain> show calls
API CALL [show(calls)] output:
created,created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid
2009-06-26
02:16:05,1245968165,switch_ivr_multi_threaded_bridge,1005,1005,inbound1000000000,sofia/external/
1005 at uat.pbx.starpoundtech.net
,4fa86434-b542-4066-99af-5924c78ddab7,1005,1005,
1000000000 at 192.168.147.130:5060,sofia/external/
1000000000 at 192.168.147.130:5060,73df8735-fee2-464d-aec0-fda886ba2cba
2009-06-26
02:16:07,1245968167,switch_ivr_multi_threaded_bridge,1005,1005,1001,sofia/external/
1005 at 192.168.147.1
,1c2c5f6d-669f-4432-ad04-35a64dbc8a14,1005,1005,sip:1001 at 192.168.147.1:5060
;fs_nat=yes,sofia/doublenat5090/sip:1001 at 192.168.147.1:5060
;fs_nat=yes,66895f68-70bf-410a-bff7-cda9549c102d
2 total.
freeswitch at localhost.localdomain> 2009-06-26 02:18:09 [DEBUG] sofia.c:2728
sofia_handle_sip_i_state() Channel sofia/external/
1005 at uat.pbx.starpoundtech.net entering state [calling]
2009-06-26 02:18:10 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready]
2009-06-26 02:18:10 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=- 1 3 IN IP4 192.168.147.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.147.1
t=0 0
m=audio 47590 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2009-06-26 02:19:07 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1005 at uat.pbx.starpoundtech.net entering state [calling]
2009-06-26 02:19:08 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel
sofia/external/1005 at uat.pbx.starpoundtech.net entering state [ready]
2009-06-26 02:19:08 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote
SDP:
v=0
o=- 1 3 IN IP4 192.168.147.1
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.147.1
t=0 0
m=audio 47590 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Artem
On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov <shiyanov at gmail.com> wrote:
> Hello!
>
> I got a problem with one way audio, symptoms are:
> firstly play audio file to channel A (A is hears sound)
> secondly bridge channel B with A (A doesn't hear B).
>
> Environment:
> - no NAT
> - User Agents being used X-Lite, EyeBeam, SJphone - same result for all of
> them- no audio, Wireshark shows that there is no RTP-flow to A from
> FreeSwitch
> - dialplan:
> <extension name="playback_media_file">
> <condition field="destination_number" expression="playmedia">
> <action application="answer"/>
> <action application="playback" data="test.wav"/>
> </condition>
> </extension>
>
> <extension name="Local_Extension_from_SP">
> <condition field="destination_number" expression="^([0-9]{2,9})$">
> <action application="set" data="dialed_extension=$1"/>
> <action application="export" data="dialed_extension=$1"/>
> </condition>
> <condition field="${sip_to_host}" expression="^([^.]*)\..*$">
> <action application="set" data="orgname=$1"/>
> </condition>
> <condition field="destination_number"
> expression="^${caller_id_number}$">
> <anti-action application="set" data="ringback=${us-ring}"/>
> <anti-action application="set"
> data="transfer_ringback=${us-ring}"/>
> <anti-action application="set" data="call_timeout=10"/>
> <anti-action application="set" data="hangup_after_bridge=true"/>
> <anti-action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED"/>
>
> <anti-action application="set" data="continue_on_fail=true"/>
> <anti-action application="db"
> data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
> <anti-action application="db"
> data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
> <anti-action application="set"
> data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)}"/>
> <anti-action application="db"
> data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
> <anti-action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
> <anti-action application="answer"/>
> <anti-action application="export"
> data="sip_h_X-SPFrom="e;${sip_from_user}"e;<${sip_from_uri}>"/>
> <anti-action application="export"
> data="sip_h_X-SPTo=<${sip_to_uri}>"/>
> <anti-action application="export"
> data="sip_h_X-SPCallId=${sip_call_id}"/>
> <anti-action application="bridge"
> data="sofia/external/${orgname}send2voicemail@
> $${starpound_sip_app_server}"/>
> </condition>
> </extension>
> - Call routing scheme:
> user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc
> Exact description what's going on is:
> user A -> FS -(bridge)-> my B2BUA
> Then my B2BUA transfers (using re-INVITE's), on behalf of user, call to
> extension "playback_media_file" . After a while B2BUA transfer (re-Inviting)
> user to extension "Local_Extension_from_SP". This should create a new call
> to user B. As a result - A doesn't hear B, but B- is OK.
> On the contrary, if a call is routed (by B2BUA) to the
> "Local_Extension_from_SP" extension (ommiting "playback_media_file" ext) -
> everything is OK.
>
>
> What I've tried:
> - set parameter "inbound-proxy-media" to "true" in Sofia profile
> - set parameter "disable_rtp_auto_adjust to "true" in Sofia profile
> Nothing helps.
>
>
> Any help or thoughts would be MUCH appreciated!
> Artem
>
>
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