[Freeswitch-users] missing 3 seconds of audio on bridge calls
Angel Carpintero
ack at telefonica.net
Fri Dec 12 17:58:01 PST 2008
Thanks again Anthony !
You fixed the issue with DTMF i had reported :
http://jira.freeswitch.org/browse/FSCORE-251
Chris Danielson added to Wiki a nice page collecting these issues with
Sonus :
http://wiki.freeswitch.org/wiki/RTP_Issues
Cheers,
El mié, 10-12-2008 a las 03:10 +0100, Angel Carpintero escribió:
> Thanks Anthony , you did a great work ! this is fixed in svn r10691.
>
> Some notes for people using Sonus and L3 as was my case :
>
> in var.xml in some scenario you may need :
>
> <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>
>
> in sip_profiles/internal.xml :
>
> <param name="rtp-rewrite-timestamps" value="true"/>
>
> might help for some people with rtp issues :
>
> <param name="rtp-timer-name" value="none"/>
>
> If you have issues with DTMF and timestamps add also :
>
> <param name="pass-rfc2833" value="true"/>
>
> I've a little issues with DTMF from VOIP , i i'll figure out can could
> be the issue , from PSTN all works like a charm :)
>
> Cheers,
>
> El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribió:
> > 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
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > > IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org
> > > iax:guest at conference.freeswitch.org/888
> > > googletalk:conf+888 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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> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:213-799-1400
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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--
Angel Carpintero
ack ( at ) telefonica ( dot ) net
Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 6EF1 B90D
"No basta saber, hay que aplicar lo que se sabe;
no basta querer hacerlas cosas, hay que hacerlas".
"Knowing is not enough; we must apply.
Willing is not enough; we must do"
Johann Wolfgang von Goethe
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