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.
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@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.
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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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:
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@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 asteriskwas not sending > the rtp on to b, and b was trying to send traffic point to point.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
--- * Olle E Johansson - oej@edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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:
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@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 topoint.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
- Olle E Johansson - oej@edvina.net
- Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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:
- handle SIP signalling NAT mangle in OpenSER
- disable the can-reinvites option in Asterisk.
- 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
Thanks for the great suggestions all.
On 11/30/06, Olle E Johansson oej@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:
- handle SIP signalling NAT mangle in OpenSER
- disable the can-reinvites option in Asterisk.
- 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
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).
rtpproxy seems to be able to handle sockets that are pointing to itself.
-ovi
On 11/30/06, Andreas Sikkema h323@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-
Andreas Sikkema -
Bogdan-Andrei Iancu -
John Peters -
Olle E Johansson -
Ovidiu Sas