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kawarod at laposte.net Guest
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Posted: Fri Mar 27, 2009 2:52 am Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 |
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Hi,
I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
versa) and it's working fine when FS stays in the media path with
default configuration.
But my setup is the following:
- Core network requires SIP INFO
- Peerings require RFC2833
all would be fine with FS if my SIP Peers were not enforcing G729
(discarding G711) so that I have to use the directive <action
application="set" data="proxy_media=true"/> in my dialplan cause FS
can't deal with G729 except in pass-through.
It's sad, but G729 is still a reality in Telco World.
So do you think there could be a way to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this, but it
sucks and that's the last step before final migration to FS.
regards,
rod
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brian at freeswitch.org Guest
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Posted: Fri Mar 27, 2009 9:26 am Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 |
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On Mar 27, 2009, at 2:40 AM, rod wrote:
Quote: | Hi,
I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
versa) and it's working fine when FS stays in the media path with
default configuration.
But my setup is the following:
- Core network requires SIP INFO
- Peerings require RFC2833
all would be fine with FS if my SIP Peers were not enforcing G729
(discarding G711) so that I have to use the directive <action
application="set" data="proxy_media=true"/> in my dialplan cause FS
can't deal with G729 except in pass-through.
|
Can't use proxy media in this case. (I highly recommend you not use Proxy Media mode)
Quote: |
It's sad, but G729 is still a reality in Telco World.
|
Coming soon!
Quote: |
So do you think there could be a way to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this, but it
sucks and that's the last step before final migration to FS.
regards,
rod |
Brian West
brian@freeswitch.org (brian@freeswitch.org)
-- Meet us a ClueCon! http://www.cluecon.com |
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anthony.minessale at g... Guest
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Posted: Fri Mar 27, 2009 10:20 am Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 |
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if you enable mod_g729 you can use freeswitch normally with that g729 codec as long
as no transcoding is enabled (same passthru concept as proxy_media_mode)
On Fri, Mar 27, 2009 at 10:07 AM, rod <kawarod@laposte.net (kawarod@laposte.net)> wrote:
Quote: | Hi Brian,
don't understand very well your advice:
--> Can't use proxy media in this case. (I highly recommend you not use
Proxy Media mode)
If i want to hide my topology network and deal with G729, I must use
proxy media ?
Why is Proxy media mode not recommended ??
regards.
rod
Brian West wrote:
Quote: |
On Mar 27, 2009, at 2:40 AM, rod wrote:
Quote: | Hi,
I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
versa) and it's working fine when FS stays in the media path with
default configuration.
But my setup is the following:
- Core network requires SIP INFO
- Peerings require RFC2833
all would be fine with FS if my SIP Peers were not enforcing G729
(discarding G711) so that I have to use the directive <action
application="set" data="proxy_media=true"/> in my dialplan cause FS
can't deal with G729 except in pass-through.
|
Can't use proxy media in this case. (I highly recommend you not use
Proxy Media mode)
Quote: |
It's sad, but G729 is still a reality in Telco World.
|
Coming soon!
Quote: |
So do you think there could be a way to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this, but it
sucks and that's the last step before final migration to FS.
regards,
rod
|
Brian West
|
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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kawarod at laposte.net Guest
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Posted: Fri Mar 27, 2009 10:22 am Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 |
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Hi Brian,
don't understand very well your advice:
--> Can't use proxy media in this case. (I highly recommend you not use
Proxy Media mode)
If i want to hide my topology network and deal with G729, I must use
proxy media ?
Why is Proxy media mode not recommended ??
regards.
rod
Brian West wrote:
Quote: |
On Mar 27, 2009, at 2:40 AM, rod wrote:
Quote: | Hi,
I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
versa) and it's working fine when FS stays in the media path with
default configuration.
But my setup is the following:
- Core network requires SIP INFO
- Peerings require RFC2833
all would be fine with FS if my SIP Peers were not enforcing G729
(discarding G711) so that I have to use the directive <action
application="set" data="proxy_media=true"/> in my dialplan cause FS
can't deal with G729 except in pass-through.
|
Can't use proxy media in this case. (I highly recommend you not use
Proxy Media mode)
Quote: |
It's sad, but G729 is still a reality in Telco World.
|
Coming soon!
Quote: |
So do you think there could be a way to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this, but it
sucks and that's the last step before final migration to FS.
regards,
rod
|
Brian West
brian@freeswitch.org <mailto:brian@freeswitch.org>
-- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/>
------------------------------------------------------------------------
_______________________________________________
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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_______________________________________________
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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kawarod at laposte.net Guest
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Posted: Fri Mar 27, 2009 11:36 am Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 |
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Hello,
I have this error when not enablig proxy_media:
2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-03-27 19:54:44 [ERR] switch_core_io.c:723
switch_core_session_write_frame() Codec G.729 decoder error!
Sure there is an option to check. Any pointers.
regards.
Anthony Minessale wrote:
Quote: | if you enable mod_g729 you can use freeswitch normally with that g729
codec as long
as no transcoding is enabled (same passthru concept as proxy_media_mode)
On Fri, Mar 27, 2009 at 10:07 AM, rod <kawarod@laposte.net
<mailto:kawarod@laposte.net>> wrote:
Hi Brian,
don't understand very well your advice:
--> Can't use proxy media in this case. (I highly recommend you
not use
Proxy Media mode)
If i want to hide my topology network and deal with G729, I must use
proxy media ?
Why is Proxy media mode not recommended ??
regards.
rod
Brian West wrote:
Quote: |
On Mar 27, 2009, at 2:40 AM, rod wrote:
Quote: | Hi,
I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
versa) and it's working fine when FS stays in the media path with
default configuration.
But my setup is the following:
- Core network requires SIP INFO
- Peerings require RFC2833
all would be fine with FS if my SIP Peers were not enforcing G729
(discarding G711) so that I have to use the directive <action
application="set" data="proxy_media=true"/> in my dialplan cause FS
can't deal with G729 except in pass-through.
|
Can't use proxy media in this case. (I highly recommend you not use
Proxy Media mode)
Quote: |
It's sad, but G729 is still a reality in Telco World.
|
Coming soon!
Quote: |
So do you think there could be a way to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this,
|
| but it
Quote: | Quote: | sucks and that's the last step before final migration to FS.
regards,
rod
|
Brian West
brian@freeswitch.org <mailto:brian@freeswitch.org>
| <mailto:brian@freeswitch.org <mailto:brian@freeswitch.org>>
<http://www.cluecon.com/>
------------------------------------------------------------------------
<mailto:Freeswitch-users@lists.freeswitch.org>
Quote: | http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
| UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Quote: | http://www.freeswitch.org
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_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
<mailto: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
<mailto:MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
<mailto:PAYPAL%3Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<mailto:sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
pstn:213-799-1400
------------------------------------------------------------------------
_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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anthony.minessale at g... Guest
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Posted: Fri Mar 27, 2009 2:08 pm Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 |
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you have to set disable-transcoding as well to avoid any transcoding situations
On Fri, Mar 27, 2009 at 11:23 AM, rod <kawarod@laposte.net (kawarod@laposte.net)> wrote:
Quote: | Hello,
I have this error when not enablig proxy_media:
2009-03-27 19:54:44 [ERR] mod_g729.c:145 switch_g729_decode() This codec
is only usable in passthrough mode!
2009-03-27 19:54:44 [ERR] switch_core_io.c:723
switch_core_session_write_frame() Codec G.729 decoder error!
Sure there is an option to check. Any pointers.
regards.
Anthony Minessale wrote:
Quote: | if you enable mod_g729 you can use freeswitch normally with that g729
codec as long
as no transcoding is enabled (same passthru concept as proxy_media_mode)
On Fri, Mar 27, 2009 at 10:07 AM, rod <kawarod@laposte.net (kawarod@laposte.net)
<mailto:kawarod@laposte.net (kawarod@laposte.net)>> wrote:
Hi Brian,
don't understand very well your advice:
--> Can't use proxy media in this case. (I highly recommend you
not use
Proxy Media mode)
If i want to hide my topology network and deal with G729, I must use
proxy media ?
Why is Proxy media mode not recommended ??
regards.
rod
Brian West wrote:
>
> On Mar 27, 2009, at 2:40 AM, rod wrote:
>
>> Hi,
>>
>> I did some tests with FS to transcode SIP INFO to RFC2833 (and vice
>> versa) and it's working fine when FS stays in the media path with
>> default configuration.
>>
>> But my setup is the following:
>> - Core network requires SIP INFO
>> - Peerings require RFC2833
>>
>> all would be fine with FS if my SIP Peers were not enforcing G729
>> (discarding G711) so that I have to use the directive <action
>> application="set" data="proxy_media=true"/> in my dialplan cause FS
>> can't deal with G729 except in pass-through.
>
> Can't use proxy media in this case. (I highly recommend you not use
> Proxy Media mode)
>
>>
>> It's sad, but G729 is still a reality in Telco World.
>
> Coming soon!
>
>>
>> So do you think there could be a way to deal with DTMF even if not
>> analyzing RTP for transcoding. My commercial SBC is doing this,
but it
>> sucks and that's the last step before final migration to FS.
>>
>> regards,
>> rod
>
> Brian West
> brian@freeswitch.org (brian@freeswitch.org) <mailto:brian@freeswitch.org (brian@freeswitch.org)>
<mailto:brian@freeswitch.org (brian@freeswitch.org) <mailto:brian@freeswitch.org (brian@freeswitch.org)>>
>
> -- Meet us a ClueCon! http://www.cluecon.com
<http://www.cluecon.com/>
>
>
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
<mailto:Freeswitch-users@lists.freeswitch.org (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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_______________________________________________
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Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
<mailto:Freeswitch-users@lists.freeswitch.org (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 ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
<mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
<mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
<mailto:sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])>
pstn:213-799-1400
------------------------------------------------------------------------
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400 |
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