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sameer at hostnsoft.com
Guest
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 Posted: Wed Jul 09, 2014 4:56 am Post subject: [asterisk-users] switching from simple_bridge technology to |
 |
|
Hi,
with canreinvite=no and directmedia=no I and getting the message in the logs for all calls
"switching from simple_bridge technology to native_rtp"
-- Executing [102@mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-00000018 is ringing
-- SIP/102-00000018 answered SIP/101-00000017
-- Channel SIP/101-00000017 joined 'simple_bridge' basic-bridge <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from simple_bridge technology to native_rtp
> 0x7f427c068a10 -- Probation passed - setting RTP source address to 111.118.250.236:49344
> 0x7f427c068a10 -- Probation passed - setting RTP source address to 111.118.250.236:49344
> 0x7f42500168d0 -- Probation passed - setting RTP source address to 111.118.250.236:26326
> 0x7f42500168d0 -- Probation passed - setting RTP source address to 111.118.250.236:26326
-- Channel SIP/101-00000017 left 'native_rtp' basic-bridge <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 left 'native_rtp' basic-bridge <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
== Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-00000017'
I cannot understand why asterisk state diff bridges if all works same
please can anyone explain me the working bridging concept and how to configure and use bridges to route the rtp externally form asterisk.
--
Regards
Sameer Rathod8109413462 |
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mjordan at digium.com
Guest
|
| Posted: Wed Jul 09, 2014 8:07 am Post subject: [asterisk-users] switching from simple_bridge technology to |
|
|
On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod <sameer@hostnsoft.com> wrote:
| Quote: | Hi,
with canreinvite=no and directmedia=no I and getting the message in the logs
for all calls
"switching from simple_bridge technology to native_rtp"
-- Executing [102@mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-00000018 is ringing
-- SIP/102-00000018 answered SIP/101-00000017
-- Channel SIP/101-00000017 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
| Quote: | Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
|
simple_bridge technology to native_rtp
| Quote: | 0x7f427c068a10 -- Probation passed - setting RTP source address to
|
111.118.250.236:49344
| Quote: | 0x7f427c068a10 -- Probation passed - setting RTP source address to
|
111.118.250.236:49344
| Quote: | 0x7f42500168d0 -- Probation passed - setting RTP source address to
|
111.118.250.236:26326
| Quote: | 0x7f42500168d0 -- Probation passed - setting RTP source address to
|
111.118.250.236:26326
-- Channel SIP/101-00000017 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
== Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-00000017'
I cannot understand why asterisk state diff bridges if all works same
please can anyone explain me the working bridging concept and how to
configure and use bridges to route the rtp externally form asterisk.
|
I think I just answered this in your other thread, but I'll repeat it here.
First, canreinvite has been deprecated as a naming convention for ...
a long time. It's not even documented any more. The code will accept
it, but all you're doing is setting the directmedia option twice:
} else if (!strcasecmp(v->name, "directmedia") ||
!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
The native RTP bridge in Asterisk 12 manages bridges between two RTP
capable channels. The bridge can either be formed remotely (in which
case the media flows between the endpoints) or locally, in which case
the media is swapped across the ports. It will attempt to perform a
remote bridge if possible, while falling back to a local bridge if a
remote bridge is not possible.
In your particular case, you've explicitly told it to *not* do
directmedia. So it won't perform a remote bridge.
Even if you set directmedia=yes (or one of its variants), you may not
have a successful remote bridge if one of the endpoints is behind a
NAT. The sip.conf sample configuration documentation is actually quite
good on this subject:
;----------------------------------- MEDIA HANDLING
--------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal
path. If there's
; no reason for Asterisk to stay in the media path, the media will be
redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set
directmedia=nonat.
;
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of
them is behind a NAT).
; The default setting is YES. If you
have all clients
; behind a NAT, or for some other
reason want Asterisk to
; stay in the audio path, you may want
to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4
- setting up the
; call directly between the endpoints
instead of sending
; a re-INVITE).
; Additionally this option does not
disable all reINVITE operations.
; It only controls Asterisk generating
reINVITEs for the specific
; purpose of setting up a direct media
path. If a reINVITE is
; needed to switch a media stream to
inactive (when placed on
; hold) or to T.38, it will still be
done, regardless of this
; setting. Note that direct T.38 is
not supported.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
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sameer at hostnsoft.com
Guest
|
| Posted: Thu Jul 10, 2014 4:28 am Post subject: [asterisk-users] switching from simple_bridge technology to |
|
|
Hi Matt,
I also tested the directmedia=yes over 3g connection ie with a public ip but I am getting only one way audio
am I doing anything wrong?
On Wed, Jul 9, 2014 at 6:54 PM, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:
| Quote: | Hi Matt,
Thank you so much for explaining me this concept
One more thing when I did testing for the above in different cases ie with directmedia=yes and no I got the flow of packets attached with this mail
Please have a look
The flow stats that the rtp packet flows directly between end point
So as per above details probably it is due to both of my endpoints are on the same network ie one side of the nat
am i right?
On Wed, Jul 9, 2014 at 6:36 PM, Matthew Jordan <mjordan@digium.com (mjordan@digium.com)> wrote:
| Quote: | On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:
| Quote: | Hi,
with canreinvite=no and directmedia=no I and getting the message in the logs
for all calls
"switching from simple_bridge technology to native_rtp"
-- Executing [102@mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-00000018 is ringing
-- SIP/102-00000018 answered SIP/101-00000017
-- Channel SIP/101-00000017 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
simple_bridge technology to native_rtp
> 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
> 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
> 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
> 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
-- Channel SIP/101-00000017 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
== Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-00000017'
I cannot understand why asterisk state diff bridges if all works same
please can anyone explain me the working bridging concept and how to
configure and use bridges to route the rtp externally form asterisk.
|
I think I just answered this in your other thread, but I'll repeat it here.
First, canreinvite has been deprecated as a naming convention for ...
a long time. It's not even documented any more. The code will accept
it, but all you're doing is setting the directmedia option twice:
} else if (!strcasecmp(v->name, "directmedia") ||
!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
The native RTP bridge in Asterisk 12 manages bridges between two RTP
capable channels. The bridge can either be formed remotely (in which
case the media flows between the endpoints) or locally, in which case
the media is swapped across the ports. It will attempt to perform a
remote bridge if possible, while falling back to a local bridge if a
remote bridge is not possible.
In your particular case, you've explicitly told it to *not* do
directmedia. So it won't perform a remote bridge.
Even if you set directmedia=yes (or one of its variants), you may not
have a successful remote bridge if one of the endpoints is behind a
NAT. The sip.conf sample configuration documentation is actually quite
good on this subject:
;----------------------------------- MEDIA HANDLING
--------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal
path. If there's
; no reason for Asterisk to stay in the media path, the media will be
redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set
directmedia=nonat.
;
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of
them is behind a NAT).
; The default setting is YES. If you
have all clients
; behind a NAT, or for some other
reason want Asterisk to
; stay in the audio path, you may want
to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4
- setting up the
; call directly between the endpoints
instead of sending
; a re-INVITE).
; Additionally this option does not
disable all reINVITE operations.
; It only controls Asterisk generating
reINVITEs for the specific
; purpose of setting up a direct media
path. If a reINVITE is
; needed to switch a media stream to
inactive (when placed on
; hold) or to T.38, it will still be
done, regardless of this
; setting. Note that direct T.38 is
not supported.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
Regards
Sameer Rathod8109413462
|
--
Regards
Sameer Rathod8109413462 |
|
| Back to top |
|
|
sameer at hostnsoft.com
Guest
|
| Posted: Thu Jul 10, 2014 4:30 am Post subject: [asterisk-users] switching from simple_bridge technology to |
|
|
sorry I forgot to add the conf
sip.conf
[101]
type=friend
username=101
secret=101
host=dynamic
context=mkg
;nat=force_rport,comedia
;dtmfmode=rfc2833
;canreinvite=no
directmedia=yes
;directrtpsetup=yes
;avpf=yes
;encryption=yes
;disallow=all
;allow=ulaw
;icesupport=yes
[102]
type=friend
username=102
secret=101
host=dynamic
context=mkg
;nat=force_rport,comedia
;dtmfmode=rfc2833
;canreinvite=no
directmedia=yes
;directrtpsetup=yes
;avpf=yes
;encryption=yes
;disallow=all
;allow=ulaw
;icesupport=yes
On Thu, Jul 10, 2014 at 2:58 PM, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:
| Quote: | Hi Matt,
I also tested the directmedia=yes over 3g connection ie with a public ip but I am getting only one way audio
am I doing anything wrong?
On Wed, Jul 9, 2014 at 6:54 PM, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:
| Quote: | Hi Matt,
Thank you so much for explaining me this concept
One more thing when I did testing for the above in different cases ie with directmedia=yes and no I got the flow of packets attached with this mail
Please have a look
The flow stats that the rtp packet flows directly between end point
So as per above details probably it is due to both of my endpoints are on the same network ie one side of the nat
am i right?
On Wed, Jul 9, 2014 at 6:36 PM, Matthew Jordan <mjordan@digium.com (mjordan@digium.com)> wrote:
| Quote: | On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:
| Quote: | Hi,
with canreinvite=no and directmedia=no I and getting the message in the logs
for all calls
"switching from simple_bridge technology to native_rtp"
-- Executing [102@mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-00000018 is ringing
-- SIP/102-00000018 answered SIP/101-00000017
-- Channel SIP/101-00000017 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 joined 'simple_bridge' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
simple_bridge technology to native_rtp
> 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
> 0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
> 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
> 0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
-- Channel SIP/101-00000017 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
-- Channel SIP/102-00000018 left 'native_rtp' basic-bridge
<0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
== Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-00000017'
I cannot understand why asterisk state diff bridges if all works same
please can anyone explain me the working bridging concept and how to
configure and use bridges to route the rtp externally form asterisk.
|
I think I just answered this in your other thread, but I'll repeat it here.
First, canreinvite has been deprecated as a naming convention for ...
a long time. It's not even documented any more. The code will accept
it, but all you're doing is setting the directmedia option twice:
} else if (!strcasecmp(v->name, "directmedia") ||
!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
The native RTP bridge in Asterisk 12 manages bridges between two RTP
capable channels. The bridge can either be formed remotely (in which
case the media flows between the endpoints) or locally, in which case
the media is swapped across the ports. It will attempt to perform a
remote bridge if possible, while falling back to a local bridge if a
remote bridge is not possible.
In your particular case, you've explicitly told it to *not* do
directmedia. So it won't perform a remote bridge.
Even if you set directmedia=yes (or one of its variants), you may not
have a successful remote bridge if one of the endpoints is behind a
NAT. The sip.conf sample configuration documentation is actually quite
good on this subject:
;----------------------------------- MEDIA HANDLING
--------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal
path. If there's
; no reason for Asterisk to stay in the media path, the media will be
redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set
directmedia=nonat.
;
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of
them is behind a NAT).
; The default setting is YES. If you
have all clients
; behind a NAT, or for some other
reason want Asterisk to
; stay in the audio path, you may want
to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4
- setting up the
; call directly between the endpoints
instead of sending
; a re-INVITE).
; Additionally this option does not
disable all reINVITE operations.
; It only controls Asterisk generating
reINVITEs for the specific
; purpose of setting up a direct media
path. If a reINVITE is
; needed to switch a media stream to
inactive (when placed on
; hold) or to T.38, it will still be
done, regardless of this
; setting. Note that direct T.38 is
not supported.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
Regards
Sameer Rathod8109413462
|
--
Regards
Sameer Rathod8109413462
|
--
Regards
Sameer Rathod8109413462 |
|
| Back to top |
|
|
mjordan at digium.com
Guest
|
| Posted: Thu Jul 10, 2014 6:49 am Post subject: [asterisk-users] switching from simple_bridge technology to |
|
|
On Thu, Jul 10, 2014 at 4:28 AM, Sameer Rathod <sameer@hostnsoft.com> wrote:
| Quote: | Hi Matt,
I also tested the directmedia=yes over 3g connection ie with a public ip but
I am getting only one way audio
am I doing anything wrong?
|
If you are getting one way audio when direct media is enabled, then
one of the devices cannot find the other. This is most likely because
one of the devices is behind a NAT.
If both devices are truly publicly accessible, then you would have to
look at a pcap at the re-INVITEs sent to the devices to determine
where Asterisk told the devices to send their media, then debug your
network to determine why media could not be sent directly between
those two devices.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
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