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ack at telefonica.net
Guest
|
 Posted: Tue Dec 09, 2008 9:14 pm Post subject: [Freeswitch-users] missing 3 seconds of audio on bridge call |
 |
|
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:
| Quote: | 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@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:
| Quote: | 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@telefonica.net>
| Quote: | wrote:
Hi guys,
I've a strange issue with FS , version svn
|
-r10584 ,
| Quote: | when FS bridges a call first 3 seconds of audio are
|
missing ,
| Quote: | looks that
only happens on PSTN calls and using ringback or
transfer_ringback. This
only happens in calls from PSTN , not from VOIP.
|
Some
| Quote: | scenarios i tried
to isolate this issue :
- Issue
PSTN --> FS ( brigde ) ringback/transfer_ringback ->
|
PSTN
| Quote: |
- Good setting bypass_media before run bridge but i
|
need rtp
| Quote: | in FS path
PSTN --> FS ( brigde ) ringback/transfer_ringback ->
|
PSTN
| Quote: |
- Good
PSTN --> FS ( brigde ) WITHOUT
|
ringback/transfer_ringback ->
| Quote: | 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|
| Quote: | blah"/>
<action application="set"
data="hangup_after_bridge=false"/>
<action application="set"
|
data="playback_terminators=#"/>
| Quote: | <action application="set" data="ringback=
|
$${us-ring}"/>
| Quote: | <action application="set"
|
data="transfer_ringback=
| Quote: | $${hold_music}"/>
<action application="set"
|
data="effective_caller_id_name=
| Quote: | ${caller_id_name}"/>
<action application="set"
data="effective_caller_id_number=
${caller_id_number}"/>
<action application="set"
|
data="originate_timeout=30"/>
| Quote: | <action application="set"
|
data="call_timeout=30"/>
| Quote: | <action application="bridge"
data="sofia/default/18008226235@PSTN_GW"/>
<action application="speak" data="cepstral|
|
Allison-8kHz|
| Quote: | 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
| Quote: | --
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
--
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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
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ack at telefonica.net
Guest
|
| Posted: Fri Dec 12, 2008 9:01 pm Post subject: [Freeswitch-users] missing 3 seconds of audio on bridge call |
|
|
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:
| Quote: | 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:
| Quote: | 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@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:
| Quote: | 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@telefonica.net>
| Quote: | wrote:
Hi guys,
I've a strange issue with FS , version svn
|
-r10584 ,
| Quote: | when FS bridges a call first 3 seconds of audio are
|
missing ,
| Quote: | looks that
only happens on PSTN calls and using ringback or
transfer_ringback. This
only happens in calls from PSTN , not from VOIP.
|
Some
| Quote: | scenarios i tried
to isolate this issue :
- Issue
PSTN --> FS ( brigde ) ringback/transfer_ringback ->
|
PSTN
| Quote: |
- Good setting bypass_media before run bridge but i
|
need rtp
| Quote: | in FS path
PSTN --> FS ( brigde ) ringback/transfer_ringback ->
|
PSTN
| Quote: |
- Good
PSTN --> FS ( brigde ) WITHOUT
|
ringback/transfer_ringback ->
| Quote: | 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|
| Quote: | blah"/>
<action application="set"
data="hangup_after_bridge=false"/>
<action application="set"
|
data="playback_terminators=#"/>
| Quote: | <action application="set" data="ringback=
|
$${us-ring}"/>
| Quote: | <action application="set"
|
data="transfer_ringback=
| Quote: | $${hold_music}"/>
<action application="set"
|
data="effective_caller_id_name=
| Quote: | ${caller_id_name}"/>
<action application="set"
data="effective_caller_id_number=
${caller_id_number}"/>
<action application="set"
|
data="originate_timeout=30"/>
| Quote: | <action application="set"
|
data="call_timeout=30"/>
| Quote: | <action application="bridge"
data="sofia/default/18008226235@PSTN_GW"/>
<action application="speak" data="cepstral|
|
Allison-8kHz|
| Quote: | 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
| Quote: | --
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
|
--
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
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org |
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