[Freeswitch-users] missing 3 seconds of audio on bridge calls

[Anthony Minessale]

most likely it's because during the time you are dong artificial ringback
the other side is not doing RTP right.

When the call is answered we flush the rtp buffer and your missing audio is
probably flushed with it.
so you can choose to have a 3 second delay or erase the 3 seconds as it does
now.

This is a known problem with sonus which has been proven to build up an
audio delay during the time
you are waiting for the call to answer.  I'm sure you prefer the way it is
to a large audio delay.



On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero <ack at telefonica.net>wrote:

> No TDM , all is SIP :
>
>
> PSTN ---> Sip Proxy_A --> FS ( brigde ) ringback/transfer_ringback
> -> Sip Proxy_B --> PSTN
>
>
> In logfile i think you can get some details about Media Gateways
> ( Sonus ) PSTN inbound / outbound is provided by Level3.
>
> I can get a capture of a call if you want, in capture the audio is not
> missing, issue with :
>
> - rtp buffer ?
> - Sonus ?
>
> Let me know anything you need so i can provide a log or create a new
> scenario.
>
>
> Thanks,
>
> El mié, 03-12-2008 a las 22:12 -0600, Anthony Minessale escribió:
> > what does PSTN represent?
> >
> > I know what the PSTN is but how are you reaching it?
> > is it TDM, SIP etc... what gateway type other details.
> >
> >
> > On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero <ack at telefonica.net>
> > wrote:
> >         Hi guys,
> >
> >          I've a strange issue with FS , version svn -r10584 ,
> >         when FS bridges a call first 3 seconds of audio are missing ,
> >         looks that
> >         only happens on PSTN calls and using ringback or
> >         transfer_ringback. This
> >         only happens in calls from PSTN , not from VOIP. Some
> >         scenarios i tried
> >         to isolate this issue :
> >
> >
> >         - Issue
> >
> >         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
> >
> >         - Good setting bypass_media before run bridge but i need rtp
> >         in FS path
> >
> >         PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN
> >
> >         - Good
> >
> >         PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback ->
> >         PSTN
> >
> >         - Always good
> >
> >         VOIP --> FS ( brigde ) -> PSTN
> >
> >
> >         Dialplan has nothing wrong ( i guess ):
> >
> >         <extension name="Transfers">
> >            <condition field="destination_number"
> >         expression="^1??XXXXXXXXXX$">
> >              <action application="answer"/>
> >              <action application="speak" data="cepstral|Allison-8kHz|
> >         blah"/>
> >              <action application="set"
> >         data="hangup_after_bridge=false"/>
> >              <action application="set" data="playback_terminators=#"/>
> >              <action application="set" data="ringback=$${us-ring}"/>
> >              <action application="set" data="transfer_ringback=
> >         $${hold_music}"/>
> >              <action application="set" data="effective_caller_id_name=
> >         ${caller_id_name}"/>
> >              <action application="set"
> >         data="effective_caller_id_number=
> >         ${caller_id_number}"/>
> >              <action application="set" data="originate_timeout=30"/>
> >              <action application="set" data="call_timeout=30"/>
> >              <action application="bridge"
> >         data="sofia/default/18008226235 at PSTN_GW"/>
> >              <action application="speak" data="cepstral|Allison-8kHz|
> >         Transfer
> >         finished"/>
> >              <action application="hangup"/>
> >            </condition>
> >          </extension>
> >
> >
> >
> >         Any ideas ?
> >
> >         Attached log of FS ( F8 from console ).
> >
> >
> >         Thanks in advance !
> >
> >         --
> >         Angel Carpintero
> >         ack ( at ) telefonica ( dot ) net
> >
> >         Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61
> >         6EF1 B90D
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
>
> --
> Angel Carpintero
> ack ( at ) telefonica ( dot ) net
>
> Key fingerprint = 3FD3 9C90 149E 7824 CECD  6BCF AC2C CA61 6EF1 B90D
>
>
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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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