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[Freeswitch-users] SIP INFO <-> RFC2833


 
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kawarod at laposte.net
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PostPosted: Fri Mar 27, 2009 2:52 am    Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 Reply with 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.

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
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PostPosted: Fri Mar 27, 2009 9:26 am    Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 Reply with 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 (brian@freeswitch.org)



-- Meet us a ClueCon! http://www.cluecon.com
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anthony.minessale at g...
Guest





PostPosted: Fri Mar 27, 2009 10:20 am    Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 Reply with 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)> 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


Quote:
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/>



------------------------------------------------------------------------

Quote:

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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])
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kawarod at laposte.net
Guest





PostPosted: Fri Mar 27, 2009 10:22 am    Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 Reply with 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
brian@freeswitch.org <mailto:brian@freeswitch.org>

-- Meet us a ClueCon! http://www.cluecon.com <http://www.cluecon.com/>



------------------------------------------------------------------------

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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


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kawarod at laposte.net
Guest





PostPosted: Fri Mar 27, 2009 11:36 am    Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 Reply with 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
<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>>
Quote:

-- Meet us a ClueCon! http://www.cluecon.com
<http://www.cluecon.com/>
------------------------------------------------------------------------
Quote:

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
<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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<mailto:Freeswitch-users@lists.freeswitch.org>
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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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anthony.minessale at g...
Guest





PostPosted: Fri Mar 27, 2009 2:08 pm    Post subject: [Freeswitch-users] SIP INFO <-> RFC2833 Reply with quote

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
    >

    _______________________________________________
    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




--
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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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])
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