[Freeswitch-users] missing 3 seconds of audio on bridge calls
Angel Carpintero
ack at telefonica.net
Tue Dec 9 18:10:10 PST 2008
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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>
>
>
>
> --
> 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
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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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