8 Dec
2020
8 Dec
'20
7:01 a.m.
Hello,
We have a problem with a SIP doorbell device which sends media one way
only, and NAT at the receiving device.
When the doorbell button is pressed it makes a call to a configured
destination. Since the doorbell only sends and doesn't receive it sends the
INVITE with sendonly in the SDP, and the destination then replies with a
200 OK with recvonly in the SDP. The problem is that the destination is
behind NAT, and its reply contains a private network IP in the SDP.
Normally Asterisk when nat=yes works around that by adjusting the
destination for RTP to be the address it actually receives audio from,
however because this device is recvonly Asterisk never receives audio from
it. This means Asterisk keeps trying to send the doorbell's RTP to the
private network IP which of course fails, and the destination never gets
the RTP from the doorbell.
We haven't found a solution in Asterisk to this, so are now looking to
Kamailio which acts as a load-balancing proxy in front of Asterisk for one.
For example, maybe we could use fix_nated_sdp, but only on 200 OK's with
recvonly.
Has anyone else encountered this, and are there any recommended solutions?
Thank you in advance!
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
8 Dec
8 Dec
8:38 a.m.
RTPEngine does the same kind of port latching that Asterisk would — that is, wait to
receive an RTP frame in order to learn the external port that the NAT gateway has mapped
the device’s (symmetric) RTP endpoint to. If it never gets one, it won’t learn. That’s the
industry-standard solution.
No magic workarounds, alas.
—
Sent from mobile, with due apologies for brevity and errors.
On Dec 7, 2020, at 11:02 PM, David Cunningham
<dcunningham(a)voisonics.com> wrote:
Hello,
We have a problem with a SIP doorbell device which sends media one way only, and NAT at
the receiving device.
When the doorbell button is pressed it makes a call to a configured destination. Since
the doorbell only sends and doesn't receive it sends the INVITE with sendonly in the
SDP, and the destination then replies with a 200 OK with recvonly in the SDP. The problem
is that the destination is behind NAT, and its reply contains a private network IP in the
SDP.
Normally Asterisk when nat=yes works around that by adjusting the destination for RTP to
be the address it actually receives audio from, however because this device is recvonly
Asterisk never receives audio from it. This means Asterisk keeps trying to send the
doorbell's RTP to the private network IP which of course fails, and the destination
never gets the RTP from the doorbell.
We haven't found a solution in Asterisk to this, so are now looking to Kamailio which
acts as a load-balancing proxy in front of Asterisk for one. For example, maybe we could
use fix_nated_sdp, but only on 200 OK's with recvonly.
Has anyone else encountered this, and are there any recommended solutions?
Thank you in advance!
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
11:56 a.m.
Hello,
if the endpoint is not behind a port forwarding nat/firewall (when one
can instruct the rtp relay to use signaling address), then probably you
can try to remove the sendonly from the INVITE SDP. That will enable rtp
from endpoint to doorbell, which may rise additional concerns (e.g.,
privacy) if its is explicitly not wanted to happen. But as Alex said in
another response, the only way to get it work is to make the endpoint
behind nat to send a RTP packet. Usually the doorbells I encountered so
far were accepting incoming traffic as well, to discuss/interact with
the person ringing on it.
Cheers,
Daniel
On 08.12.20 05:01, David Cunningham wrote:
Hello,
We have a problem with a SIP doorbell device which sends media one way
only, and NAT at the receiving device.
When the doorbell button is pressed it makes a call to a configured
destination. Since the doorbell only sends and doesn't receive it
sends the INVITE with sendonly in the SDP, and the destination then
replies with a 200 OK with recvonly in the SDP. The problem is that
the destination is behind NAT, and its reply contains a private
network IP in the SDP.
Normally Asterisk when nat=yes works around that by adjusting the
destination for RTP to be the address it actually receives audio from,
however because this device is recvonly Asterisk never receives audio
from it. This means Asterisk keeps trying to send the doorbell's RTP
to the private network IP which of course fails, and the destination
never gets the RTP from the doorbell.
We haven't found a solution in Asterisk to this, so are now looking to
Kamailio which acts as a load-balancing proxy in front of Asterisk for
one. For example, maybe we could use fix_nated_sdp, but only on 200
OK's with recvonly.
Has anyone else encountered this, and are there any recommended solutions?
Thank you in advance!
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
9 Dec
9 Dec
10:47 a.m.
Hi Daniel,
Removing the sendonly from the INVITE SDP sounds like the most workable
solution in our case. We'd only want to do it for a new INVITE though, not
a re-INVITE in a situation where a call is put on hold. Would you be able
to give an example of such a configuration?
Thanks very much for your help!
On Tue, 8 Dec 2020 at 21:56, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
if the endpoint is not behind a port forwarding nat/firewall (when one can
instruct the rtp relay to use signaling address), then probably you can try
to remove the sendonly from the INVITE SDP. That will enable rtp from
endpoint to doorbell, which may rise additional concerns (e.g., privacy) if
its is explicitly not wanted to happen. But as Alex said in another
response, the only way to get it work is to make the endpoint behind nat to
send a RTP packet. Usually the doorbells I encountered so far were
accepting incoming traffic as well, to discuss/interact with the person
ringing on it.
Cheers,
Daniel
On 08.12.20 05:01, David Cunningham wrote:
Hello,
We have a problem with a SIP doorbell device which sends media one way
only, and NAT at the receiving device.
When the doorbell button is pressed it makes a call to a configured
destination. Since the doorbell only sends and doesn't receive it sends the
INVITE with sendonly in the SDP, and the destination then replies with a
200 OK with recvonly in the SDP. The problem is that the destination is
behind NAT, and its reply contains a private network IP in the SDP.
Normally Asterisk when nat=yes works around that by adjusting the
destination for RTP to be the address it actually receives audio from,
however because this device is recvonly Asterisk never receives audio from
it. This means Asterisk keeps trying to send the doorbell's RTP to the
private network IP which of course fails, and the destination never gets
the RTP from the doorbell.
We haven't found a solution in Asterisk to this, so are now looking to
Kamailio which acts as a load-balancing proxy in front of Asterisk for one.
For example, maybe we could use fix_nated_sdp, but only on 200 OK's with
recvonly.
Has anyone else encountered this, and are there any recommended solutions?
Thank you in advance!
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda --
www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
10:49 a.m.
The salient quality of a reinvite is that has_totag() == true; it is
handled in the loose_route() section of your config. You want to do the
SDP manipulation in the part below that, where initial INVITEs are handled.
On 12/9/20 2:47 AM, David Cunningham wrote:
Hi Daniel,
Removing the sendonly from the INVITE SDP sounds like the most workable
solution in our case. We'd only want to do it for a new INVITE though,
not a re-INVITE in a situation where a call is put on hold. Would you be
able to give an example of such a configuration?
Thanks very much for your help!
On Tue, 8 Dec 2020 at 21:56, Daniel-Constantin Mierla <miconda(a)gmail.com
<mailto:miconda@gmail.com>> wrote:
Hello,
if the endpoint is not behind a port forwarding nat/firewall (when
one can instruct the rtp relay to use signaling address), then
probably you can try to remove the sendonly from the INVITE SDP.
That will enable rtp from endpoint to doorbell, which may rise
additional concerns (e.g., privacy) if its is explicitly not wanted
to happen. But as Alex said in another response, the only way to get
it work is to make the endpoint behind nat to send a RTP packet.
Usually the doorbells I encountered so far were accepting incoming
traffic as well, to discuss/interact with the person ringing on it.
Cheers,
Daniel
On 08.12.20 05:01, David Cunningham wrote:
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hello,
We have a problem with a SIP doorbell device which sends media one
way only, and NAT at the receiving device.
When the doorbell button is pressed it makes a call to a
configured destination. Since the doorbell only sends and doesn't
receive it sends the INVITE with sendonly in the SDP, and the
destination then replies with a 200 OK with recvonly in the SDP.
The problem is that the destination is behind NAT, and its reply
contains a private network IP in the SDP.
Normally Asterisk when nat=yes works around that by adjusting the
destination for RTP to be the address it actually receives audio
from, however because this device is recvonly Asterisk never
receives audio from it. This means Asterisk keeps trying to send
the doorbell's RTP to the private network IP which of course
fails, and the destination never gets the RTP from the doorbell.
We haven't found a solution in Asterisk to this, so are now
looking to Kamailio which acts as a load-balancing proxy in front
of Asterisk for one. For example, maybe we could use
fix_nated_sdp, but only on 200 OK's with recvonly.
Has anyone else encountered this, and are there any recommended
solutions?
Thank you in advance!
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
Daniel-Constantin Mierla --www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda>
--www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Funding:https://www.paypal.me/dcmierla <https://www.paypal.me/dcmierla>
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
10 Dec
10 Dec
3:43 a.m.
Hi Alex,
Thank you for that. Do you offhand know of an easy way to test if the
sendonly attribute is set? Presumably we can use
sdp_remove_line_by_prefix() to remove it.,
On Wed, 9 Dec 2020 at 20:52, Alex Balashov <abalashov(a)evaristesys.com>
wrote:
The salient quality of a reinvite is that has_totag()
== true; it is
handled in the loose_route() section of your config. You want to do the
SDP manipulation in the part below that, where initial INVITEs are handled.
On 12/9/20 2:47 AM, David Cunningham wrote:
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
Hi Daniel,
Removing the sendonly from the INVITE SDP sounds like the most workable
solution in our case. We'd only want to do it for a new INVITE though,
not a re-INVITE in a situation where a call is put on hold. Would you be
able to give an example of such a configuration?
Thanks very much for your help!
On Tue, 8 Dec 2020 at 21:56, Daniel-Constantin Mierla <miconda(a)gmail.com
<mailto:miconda@gmail.com>> wrote:
Hello,
if the endpoint is not behind a port forwarding nat/firewall (when
one can instruct the rtp relay to use signaling address), then
probably you can try to remove the sendonly from the INVITE SDP.
That will enable rtp from endpoint to doorbell, which may rise
additional concerns (e.g., privacy) if its is explicitly not wanted
to happen. But as Alex said in another response, the only way to get
it work is to make the endpoint behind nat to send a RTP packet.
Usually the doorbells I encountered so far were accepting incoming
traffic as well, to discuss/interact with the person ringing on it.
Cheers,
Daniel
On 08.12.20 05:01, David Cunningham wrote:
> Hello,
>
> We have a problem with a SIP doorbell device which sends media one
> way only, and NAT at the receiving device.
>
> When the doorbell button is pressed it makes a call to a
> configured destination. Since the doorbell only sends and doesn't
> receive it sends the INVITE with sendonly in the SDP, and the
> destination then replies with a 200 OK with recvonly in the SDP.
> The problem is that the destination is behind NAT, and its reply
> contains a private network IP in the SDP.
>
> Normally Asterisk when nat=yes works around that by adjusting the
> destination for RTP to be the address it actually receives audio
> from, however because this device is recvonly Asterisk never
> receives audio from it. This means Asterisk keeps trying to send
> the doorbell's RTP to the private network IP which of course
> fails, and the destination never gets the RTP from the doorbell.
>
> We haven't found a solution in Asterisk to this, so are now
> looking to Kamailio which acts as a load-balancing proxy in front
> of Asterisk for one. For example, maybe we could use
> fix_nated_sdp, but only on 200 OK's with recvonly.
>
> Has anyone else encountered this, and are there any recommended
> solutions?
>
> Thank you in advance!
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/ <http://voisonics.com/>
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
Daniel-Constantin Mierla --www.asipto.com <http://www.asipto.com>
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Funding:https://www.paypal.me/dcmierla <
https://www.paypal.me/dcmierla>
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:07 a.m.
if(has_body("application/sdp") && search("a=sendonly")) [1]
It's what comes to mind off the top of my head, anyway.
-- Alex
[1] https://www.kamailio.org/docs/modules/stable/modules/textops.html
On 12/9/20 7:43 PM, David Cunningham wrote:
Hi Alex,
Thank you for that. Do you offhand know of an easy way to test if the
sendonly attribute is set? Presumably we can use
sdp_remove_line_by_prefix() to remove it.,
On Wed, 9 Dec 2020 at 20:52, Alex Balashov <abalashov(a)evaristesys.com
<mailto:abalashov@evaristesys.com>> wrote:
The salient quality of a reinvite is that has_totag() == true; it is
handled in the loose_route() section of your config. You want to do the
SDP manipulation in the part below that, where initial INVITEs are
handled.
On 12/9/20 2:47 AM, David Cunningham wrote:
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/ <http://www.evaristesys.com/>,
http://www.csrpswitch.com/ <http://www.csrpswitch.com/>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hi Daniel,
Removing the sendonly from the INVITE SDP sounds like the most
workable
solution in our case. We'd only want to do it
for a new INVITE
though,
not a re-INVITE in a situation where a call is
put on hold. Would
you be
able to give an example of such a configuration?
Thanks very much for your help!
On Tue, 8 Dec 2020 at 21:56, Daniel-Constantin Mierla
<miconda(a)gmail.com
<mailto:miconda@gmail.com>
<mailto:miconda@gmail.com
<mailto:miconda@gmail.com>>> wrote:
Hello,
if the endpoint is not behind a port forwarding nat/firewall
(when
one can instruct the rtp relay to use
signaling address), then
probably you can try to remove the sendonly from the INVITE SDP.
That will enable rtp from endpoint to doorbell, which may rise
additional concerns (e.g., privacy) if its is explicitly not
wanted
to happen. But as Alex said in another
response, the only way
to get
it work is to make the endpoint behind nat to
send a RTP packet.
Usually the doorbells I encountered so far were accepting
incoming
traffic as well, to discuss/interact with the
person ringing
on it.
Cheers,
Daniel
On 08.12.20 05:01, David Cunningham wrote:
> Hello,
>
> We have a problem with a SIP doorbell device which sends
media one
> way only, and NAT at the receiving
device.
>
> When the doorbell button is pressed it makes a call to a
> configured destination. Since the doorbell only sends and
doesn't
> receive it sends the INVITE with sendonly
in the SDP, and the
> destination then replies with a 200 OK with recvonly in the SDP.
> The problem is that the destination is behind NAT, and its reply
> contains a private network IP in the SDP.
>
> Normally Asterisk when nat=yes works around that by
adjusting the
> destination for RTP to be the address it
actually receives audio
> from, however because this device is recvonly Asterisk never
> receives audio from it. This means Asterisk keeps trying to send
> the doorbell's RTP to the private network IP which of course
> fails, and the destination never gets the RTP from the doorbell.
>
> We haven't found a solution in Asterisk to this, so are now
> looking to Kamailio which acts as a load-balancing proxy in
front
> of Asterisk for one. For example, maybe
we could use
> fix_nated_sdp, but only on 200 OK's with recvonly.
>
> Has anyone else encountered this, and are there any recommended
> solutions?
>
> Thank you in advance!
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/ <http://voisonics.com/>
<http://voisonics.com/ <http://voisonics.com/>>
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>
<mailto:sr-users@lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>>
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>>
--
Daniel-Constantin Mierla --www.asipto.com
<http://www.asipto.com>
<http://www.asipto.com <http://www.asipto.com>>
www.twitter.com/miconda
<http://www.twitter.com/miconda>
<http://www.twitter.com/miconda
<http://www.twitter.com/miconda>>
--www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
<http://www.linkedin.com/in/miconda
<http://www.linkedin.com/in/miconda>>
Funding:https://www.paypal.me/dcmierla
<https://www.paypal.me/dcmierla> <https://www.paypal.me/dcmierla
<https://www.paypal.me/dcmierla>>
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
<http://voisonics.com/
<http://voisonics.com/>>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
4:17 a.m.
Thanks Alex!
On Thu, 10 Dec 2020 at 14:09, Alex Balashov <abalashov(a)evaristesys.com>
wrote:
if(has_body("application/sdp") &&
search("a=sendonly")) [1]
It's what comes to mind off the top of my head, anyway.
-- Alex
[1] https://www.kamailio.org/docs/modules/stable/modules/textops.html
On 12/9/20 7:43 PM, David Cunningham wrote:
packet.
SDP.
audio
recommended
http://www.asipto.com>>
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/ <http://www.evaristesys.com/>,
http://www.csrpswitch.com/ <http://www.csrpswitch.com/>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
David Cunningham, Voisonics Limited
http://voisonics.com/ <http://voisonics.com/>
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
Hi Alex,
Thank you for that. Do you offhand know of an easy way to test if the
sendonly attribute is set? Presumably we can use
sdp_remove_line_by_prefix() to remove it.,
On Wed, 9 Dec 2020 at 20:52, Alex Balashov <abalashov(a)evaristesys.com
<mailto:abalashov@evaristesys.com>> wrote:
The salient quality of a reinvite is that has_totag() == true; it is
handled in the loose_route() section of your config. You want to do
the
SDP manipulation in the part below that,
where initial INVITEs are
handled.
On 12/9/20 2:47 AM, David Cunningham wrote:
SDP.
Hi Daniel,
Removing the sendonly from the INVITE SDP sounds like the most
workable
solution in our case. We'd only want to do it
for a new INVITE
though,
not a re-INVITE in a situation where a call is
put on hold. Would
you be
able to give an example of such a configuration?
Thanks very much for your help!
On Tue, 8 Dec 2020 at 21:56, Daniel-Constantin Mierla
<miconda(a)gmail.com
<mailto:miconda@gmail.com>
<mailto:miconda@gmail.com
<mailto:miconda@gmail.com>>> wrote:
Hello,
if the endpoint is not behind a port forwarding nat/firewall
(when
> one can instruct the rtp relay to use signaling address), then
> probably you can try to remove the sendonly from the INVITE That
will enable rtp from endpoint to doorbell, which may rise
additional concerns (e.g., privacy) if its is explicitly not
wanted
to happen. But as Alex said in another
response, the only way
to get
> it work is to make the endpoint behind nat to send a RTP
Usually the doorbells I encountered so far were accepting
incoming
traffic as well, to discuss/interact with the
person ringing
on it.
Cheers,
Daniel
On 08.12.20 05:01, David Cunningham wrote:
> Hello,
>
> We have a problem with a SIP doorbell device which sends
media one
> way only, and NAT at the receiving
device.
>
> When the doorbell button is pressed it makes a call to a
> configured destination. Since the doorbell only sends and
doesn't
>> receive it sends the INVITE with sendonly in the SDP, and the
>> destination then replies with a 200 OK with recvonly in the >> The problem is that the
destination is behind NAT, and its
reply
>
contains a private network IP in the SDP.
>
> Normally Asterisk when nat=yes works around that by
adjusting the
>> destination for RTP to be the address it actually receives >> from, however because this
device is recvonly Asterisk never
>> receives audio from it. This means Asterisk keeps trying to
send
>> the doorbell's RTP to the
private network IP which of course
>> fails, and the destination never gets the RTP from the
doorbell.
>
> We haven't found a solution in Asterisk to this, so are now
> looking to Kamailio which acts as a load-balancing proxy in
front
>> of Asterisk for one. For example, maybe we could use
>> fix_nated_sdp, but only on 200 OK's with recvonly.
>>
>> Has anyone else encountered this, and are there any >
solutions?
>
> Thank you in advance!
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/ <http://voisonics.com/>
<http://voisonics.com/ <http://voisonics.com/>>
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>
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New Zealand: +64 (0)28 2558 3782
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8 Dec
8 Dec
8:53 p.m.
Hello David,
If the receiving NATed device has support for ICE, you can enable ICE
support on the asterisk server and then the media should flow through
(NAT pinholes are opened during ICE candidates negotiation).
Regards,
Ovidiu Sas
On Mon, Dec 7, 2020 at 11:02 PM David Cunningham
<dcunningham(a)voisonics.com> wrote:
Hello,
We have a problem with a SIP doorbell device which sends media one way only, and NAT at
the receiving device.
When the doorbell button is pressed it makes a call to a configured destination. Since
the doorbell only sends and doesn't receive it sends the INVITE with sendonly in the
SDP, and the destination then replies with a 200 OK with recvonly in the SDP. The problem
is that the destination is behind NAT, and its reply contains a private network IP in the
SDP.
Normally Asterisk when nat=yes works around that by adjusting the destination for RTP to
be the address it actually receives audio from, however because this device is recvonly
Asterisk never receives audio from it. This means Asterisk keeps trying to send the
doorbell's RTP to the private network IP which of course fails, and the destination
never gets the RTP from the doorbell.
We haven't found a solution in Asterisk to this, so are now looking to Kamailio which
acts as a load-balancing proxy in front of Asterisk for one. For example, maybe we could
use fix_nated_sdp, but only on 200 OK's with recvonly.
Has anyone else encountered this, and are there any recommended solutions?
Thank you in advance!
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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1446
days inactive
1448
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8 comments
4 participants
participants (4)
-
Alex Balashov -
Daniel-Constantin Mierla -
David Cunningham -
Ovidiu Sas