30 Nov
2006
30 Nov
'06
5:41 a.m.
Not sure why that's happening. Probably setting canreinvite=no on the
asterisk side will eliminate the re-INVITEs as a temporary solution, but
still would like to know what is happening...
wrote:
Sometimes, a calls b and b hears a, and a hears b for
a second but a
second
INVITE comes to phone B that causes it to redirect rtp
to be point to
point.
Sometimes there is no audio.
Sometimes, everything works fine.
At one point, rtp from a was going to asterisk, but
asterisk was not
sending
the rtp on to b, and b was trying to send traffic
point to point.
30 Nov
30 Nov
6:16 a.m.
ONsip has some tips for handling re-INVITEs with rtpproxy:
http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route
Advises to use force_rtp_proxy(l) on reinvites.
On 11/29/06, John Peters <petersprc(a)gmail.com> wrote:
Not sure why that's happening. Probably setting canreinvite=no on the
asterisk side will eliminate the re-INVITEs as a temporary solution, but
still would like to know what is happening...
wrote:
Sometimes, a calls b and b hears a, and a hears b
for a second but a
second
INVITE comes to phone B that causes it to
redirect rtp to be point to
point.
Sometimes there is no audio.
Sometimes, everything works fine.
At one point, rtp from a was going to asterisk,
but asterisk was not
sending
> the rtp on to b, and b was trying to send traffic point to point. 
10:42 a.m.
Hi John,
actually if you use Asterisk, there is no need for using RTPproxy as
Asterisk is able to cope with nated rtp by itslef (using Comedia).
regards,
bogdan
John Peters wrote:
ONsip has some tips for handling re-INVITEs with
rtpproxy:
http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route
<http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route>
Advises to use force_rtp_proxy(l) on reinvites.
On 11/29/06, *John Peters* <petersprc(a)gmail.com
<mailto:petersprc@gmail.com>> wrote:
Not sure why that's happening. Probably setting canreinvite=no on
the asterisk side will eliminate the re-INVITEs as a temporary
solution, but still would like to know what is happening...
wrote:
Sometimes, a calls b and b hears a, and a hears b
for a second
but a second
INVITE comes to phone B that causes it to
redirect rtp to be
point to point.
Sometimes there is no audio.
Sometimes, everything works fine.
At one point, rtp from a was going to asterisk,
but asterisk was
not sending
the rtp on to b, and b was trying to send traffic
point to point.
------------------------------------------------------------------------
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
10:45 a.m.
30 nov 2006 kl. 10.42 skrev Bogdan-Andrei Iancu:
Hi John,
actually if you use Asterisk, there is no need for using RTPproxy
as Asterisk is able to cope with nated rtp by itslef (using Comedia).
Depends on the setup really, Bogdan.
If your devices are registering with OpenSER, you need RTP proxy. If
they're registering with Asterisk and calling through
Asterisk, asterisk can handle media and NAT.
If you have NATs, you should really disable can-reinvites since you
don't want ASterisk to set up media stream
that will fail.
/O
regards,
bogdan
John Peters wrote:
---
* Olle E Johansson - oej(a)edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
ONsip has some tips for handling re-INVITEs with
rtpproxy:
http://siprouter.onsip.org/doc/gettingstarted/
ch08s02.html#rtp_loose_route <http://siprouter.onsip.org/doc/
gettingstarted/ch08s02.html#rtp_loose_route>
Advises to use force_rtp_proxy(l) on reinvites.
On 11/29/06, *John Peters* <petersprc(a)gmail.com
<mailto:petersprc@gmail.com>> wrote:
Not sure why that's happening. Probably setting canreinvite=no on
the asterisk side will eliminate the re-INVITEs as a temporary
solution, but still would like to know what is happening...
wrote:
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users Sometimes, a calls b and b hears a, and a hears b
for a second
but a second
INVITE comes to phone B that causes it to
redirect rtp to be
point to point.
Sometimes there is no audio.
Sometimes, everything works fine.
At one point, rtp from a was going to asterisk,
but asterisk
was
not sending
the rtp on to b, and b was trying to send traffic
point to
point.
---------------------------------------------------------------------
---
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
11 a.m.
Hi Olle,
even if the devices register with OpenSER and you use Asterisk as B2BUA
behind the proxy, it still works if:
1) handle SIP signalling NAT mangle in OpenSER
2) disable the can-reinvites option in Asterisk.
3) do not use rtpproxy.
From Asterisk perspective, do you see any issues in this scenario?
thanks and regards,
bogdan
Olle E Johansson wrote:
30 nov 2006 kl. 10.42 skrev Bogdan-Andrei Iancu:
Hi John,
actually if you use Asterisk, there is no need for using RTPproxy as
Asterisk is able to cope with nated rtp by itslef (using Comedia).
Depends on the setup really, Bogdan.
If your devices are registering with OpenSER, you need RTP proxy. If
they're registering with Asterisk and calling through
Asterisk, asterisk can handle media and NAT.
If you have NATs, you should really disable can-reinvites since you
don't want ASterisk to set up media stream
that will fail.
/O
regards,
bogdan
John Peters wrote:
---
* Olle E Johansson - oej(a)edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
ONsip has some tips for handling re-INVITEs with
rtpproxy:
http://siprouter.onsip.org/doc/gettingstarted/
ch08s02.html#rtp_loose_route <http://siprouter.onsip.org/doc/
gettingstarted/ch08s02.html#rtp_loose_route>
Advises to use force_rtp_proxy(l) on reinvites.
On 11/29/06, *John Peters* <petersprc(a)gmail.com
<mailto:petersprc@gmail.com>> wrote:
Not sure why that's happening. Probably setting canreinvite=no on
the asterisk side will eliminate the re-INVITEs as a temporary
solution, but still would like to know what is happening...
wrote:
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users Sometimes, a calls b and b hears a, and a hears b
for a second
but a second
INVITE comes to phone B that causes it to
redirect rtp to be
point to point.
Sometimes there is no audio.
Sometimes, everything works fine.
At one point, rtp from a was going to asterisk,
but asterisk was
not sending
the rtp on to b, and b was trying to send traffic
point to
point.
---------------------------------------------------------------------
---
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
11:03 a.m.
30 nov 2006 kl. 11.00 skrev Bogdan-Andrei Iancu:
Hi Olle,
even if the devices register with OpenSER and you use Asterisk as
B2BUA behind the proxy, it still works if:
1) handle SIP signalling NAT mangle in OpenSER
2) disable the can-reinvites option in Asterisk.
3) do not use rtpproxy.
From Asterisk perspective, do you see any issues in this scenario?
You need to add nat=yes in Asterisk to turn on Symmetric RTP. The
problem then is that Asterisk will
also enable symmetric SIP, which might hurt communication when using
a proxy.
The rtp-proxy is only needed for devices that call between each other
and need nat support.
For calls to and from Asterisk, yes, Asterisk can handle most of it
in the way you describe it.
/O
9:02 p.m.
On Thursday 30 November 2006 10:45, Olle E Johansson wrote:
If you have NATs, you should really disable
can-reinvites since you
don't want ASterisk to set up media stream
that will fail.
A year and a half ago I tried to get ser + mediaproxy + asterisk to work with
reinvite. I got it to work with a little massaging in mediaproxy, but ran
into trouble when both CPEs involved in the call were behind NAT and using
the same mediaproxy. Mediaproxy got confused with all those sockets that were
pointing to itself. I tried to solve this but got hopelessly lost in the code
(no Python experience was probably the issue).
--
Andreas Sikkema
9:07 p.m.
rtpproxy seems to be able to handle sockets that are pointing to itself.
-ovi
On 11/30/06, Andreas Sikkema <h323(a)ramdyne.nl> wrote:
On Thursday 30 November 2006 10:45, Olle E Johansson
wrote:
If you have NATs, you should really disable
can-reinvites since you
don't want ASterisk to set up media stream
that will fail.
A year and a half ago I tried to get ser + mediaproxy + asterisk to work with
reinvite. I got it to work with a little massaging in mediaproxy, but ran
into trouble when both CPEs involved in the call were behind NAT and using
the same mediaproxy. Mediaproxy got confused with all those sockets that were
pointing to itself. I tried to solve this but got hopelessly lost in the code
(no Python experience was probably the issue).
--
Andreas Sikkema
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
2 Dec
2 Dec
6:22 a.m.
Thanks for the great suggestions all.
On 11/30/06, Olle E Johansson <oej(a)edvina.net> wrote:
30 nov 2006 kl. 11.00 skrev Bogdan-Andrei Iancu:
Hi Olle,
even if the devices register with OpenSER and you use Asterisk as
B2BUA behind the proxy, it still works if:
1) handle SIP signalling NAT mangle in OpenSER
2) disable the can-reinvites option in Asterisk.
3) do not use rtpproxy.
From Asterisk perspective, do you see any issues in this scenario?
You need to add nat=yes in Asterisk to turn on Symmetric RTP. The
problem then is that Asterisk will
also enable symmetric SIP, which might hurt communication when using
a proxy.
The rtp-proxy is only needed for devices that call between each other
and need nat support.
For calls to and from Asterisk, yes, Asterisk can handle most of it
in the way you describe it.
/O
6378
days inactive
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8 comments
5 participants
participants (5)
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Andreas Sikkema -
Bogdan-Andrei Iancu -
John Peters -
Olle E Johansson -
Ovidiu Sas