[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
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>         > 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
> 
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> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
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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"

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