[Freeswitch-users] Problem: re-invite with 'inactive' SDP and 'bridge' function

Wed May 27 05:38:16 PDT 2009

Hi everyone!

In my environment I use FreeSwitch as media server and session border
controller. SIP routing is mostly done with my private B2BUA. The problem
itself is in my "hold" functionality. In details: A is calling to B:

<!--
     if the calling party is the called party, go to their VM
     if the calling party is NOT the called party dial the extension
     (1000-1019) for 30 seconds and go to voicemail if the
     call fails (continue_on_fail=true), otherwise hang up after a
     successful bridge (hangup_after-bridge=true)
-->
<extension name="Local_Extension_from_SP">
  <condition field="${sip_contact_uri}"
expression="^.*StarPound_.*_B2BUA.*$"/>
  <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}@
${domain_name}"/>
    <anti-action application="answer"/>
    <anti-action application="export"
data="sip_h_X-SPFrom=&quote;${sip_from_user}&quote;&lt;${sip_from_uri}&gt;"/>
    <anti-action application="export"
data="sip_h_X-SPTo=&lt;${sip_to_uri}&gt;"/>
    <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 being answered. Then B wants to hold a call (A should hear
music-on-hold, B should hear nothing but still have a call). To accomplish
this I:
1. on behalf of B2BUA make call to FreeSwitch "hold" extension, get SDP for
music-on-hold (moh_sdp)
<extension name="hold">
    <condition field="destination_number" expression="hold">
      <action application="answer"/>
      <action application="playback" data="$${hold_music}"/>
    </condition>
</extension>
2. re-invite A with SDP moh_sdp
3. re-invite B with it's own sdp and attribute 'inactive' (also tried
re-invite with A's SDP and 'inactive' and with 0.0.0.0 IP in SDP and
'inactive' - same results)

Result: both A and B hear music-on-hold.
I suspect that the reason is in 'bridge' function.. it seems, that it still
bridges media against the SIP.

Here is a snippet from FreeSwitch log related to "strange" (from my point of
view) behaviour - when FreeSwitch change media for B to the incorrect one.
Here 1004 is B.

2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2939 sofia_glue_negotiate_sdp()
Audio Codec Compare [GSM:3:8000:20]/[GSM:3:8000:20]
2009-05-26 13:19:14 [DEBUG] sofia_glue.c:1863 sofia_glue_tech_set_codec()
Already using GSM
2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2899 sofia_glue_negotiate_sdp()
Set 2833 dtmf payload to 101
2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2908 sofia_glue_negotiate_sdp()
Set comfort noise payload to 13
2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2139 sofia_glue_activate_rtp()
Audio params changed for sofia/external/6782886802 at 10.249.194.9:5060 from
75.101.166.29:27426 to 75.101.166.29:22922
2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2146 sofia_glue_activate_rtp()
AUDIO RTP [sofia/external/6782886802 at 10.249.194.9:5060] 10.249.194.9 port
28594 -> 75.101.166.29 port 22922 codec: 3 ms: 20
2009-05-26 13:19:14 [DEBUG] sofia_glue.c:2165 sofia_glue_activate_rtp()
AUDIO RTP CHANGING DEST TO: [75.101.166.29:22922]
2009-05-26 13:19:14 [DEBUG] sofia.c:2996 sofia_handle_sip_i_state()
Channel sofia/external/1005 at 10.249.194.9 entering state [ready][200]
2009-05-26 13:19:15 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
sofia/doublenat5090/sip:1004 at 172.16.0.6 <sip%3A1004 at 172.16.0.6>
;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060
Command Execute playback(local_stream://moh)
EXECUTE sofia/doublenat5090/sip:1004 at 172.16.0.6
<sip%3A1004 at 172.16.0.6>;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060
playback(local_stream://moh)
2009-05-26 13:19:15 [DEBUG] mod_local_stream.c:346
local_stream_file_open() Opening Stream [moh/8000] 8000hz
2009-05-26 13:19:15 [DEBUG] switch_ivr_play_say.c:1084
switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
2009-05-26 13:19:15 [DEBUG] switch_core_io.c:649
switch_core_session_write_frame()
sofia/doublenat5090/sip:1004 at 172.16.0.6 <sip%3A1004 at 172.16.0.6>
;fs_nat=yes;fs_path=sip%3A1004%40193.33.170.63%3A5060
receive message [TRANSCODING_NECESSARY]

Any thoughts would be really appreciated!
Artem
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