3 Jun
2018
3 Jun
'18
2:55 p.m.
Hello group,
The call flow is:
useragent (TLS) -> Kamailio -> Sems Server.
During the establishment of a call (before the 200OK or during a
retransmission of the 200OK issued by Sems Server via Kamailio),
if a user agent (TCP) loses its connection (the user agent is on a mobile
network for example or the network is not reliable.) and sends a "register"
to Kamailio (same source IP address but a different source TCP port), is it
possible for Kamailio to take into account this new TCP port with the
function t_relay () ?
Abdoul
4 Jun
4 Jun
2:47 p.m.
Not sure if I understood correctly your problem, but that sounds a lot like
forking the call as soon as another REGISTER comes in for the same AoR.
In that case, it may be worth to have a look into TSILO module, it may help
you in your case.
On Sun, Jun 3, 2018 at 1:55 PM, Abdoul Osséni <abdoul.osseni(a)gmail.com>
wrote:
Hello group,
The call flow is:
useragent (TLS) -> Kamailio -> Sems Server.
During the establishment of a call (before the 200OK or during a
retransmission of the 200OK issued by Sems Server via Kamailio),
if a user agent (TCP) loses its connection (the user agent is on a mobile
network for example or the network is not reliable.) and sends a
"register" to Kamailio (same source IP address but a different source TCP
port), is it possible for Kamailio to take into account this new TCP port
with the function t_relay () ?
Abdoul
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4:39 p.m.
Could this module work in scenario like in case if the call is established
and running,
And one of the UA'a lost its connection and then reconnect and send a new
register
Could the call be continued ?
Without sending new INVITE.. just like the whatsapp
On Mon, Jun 4, 2018, 2:47 PM Alfonso Pinto <alfonso.pinto(a)gmail.com> wrote:
Not sure if I understood correctly your problem, but
that sounds a lot
like forking the call as soon as another REGISTER comes in for the same AoR.
In that case, it may be worth to have a look into TSILO module, it may
help you in your case.
On Sun, Jun 3, 2018 at 1:55 PM, Abdoul Osséni <abdoul.osseni(a)gmail.com>
wrote:
Hello group,
The call flow is:
useragent (TLS) -> Kamailio -> Sems Server.
During the establishment of a call (before the 200OK or during a
retransmission of the 200OK issued by Sems Server via Kamailio),
if a user agent (TCP) loses its connection (the user agent is on a mobile
network for example or the network is not reliable.) and sends a
"register" to Kamailio (same source IP address but a different source
TCP port), is it possible for Kamailio to take into account this new TCP
port with the function t_relay () ?
Abdoul
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
5:16 p.m.
On 06/04/2018 03:39 PM, Amar Tinawi wrote:
Could this module work in scenario like in case if the
call is
established and running,
And one of the UA'a lost its connection and then reconnect and send a
new register
Could the call be continued ?
Without sending new INVITE.. just like the whatsapp
I might be wrong, but I think that GRUU and SIP-outbound RFC are
supposed to address this to some extent, because when receiving
in-dialog request the proxy can look up the new connection to the UA
using the device id and replace the connection. Not sure how this looks
like in practice and with TLS connections, since not many clients
support GRUU and some wouldn't even re-register after TCP disconnect.\
As to original question, I doubt if there is such a method for
transaction in progress, but see perhaps if topci
https://lists.cs.columbia.edu/pipermail/sip-implementors/2012-June/028480.h…
applies to you.
BR,
Andrew
8:40 p.m.
Thanks Andrew
Even if i don't Use tls ?
On Mon, Jun 4, 2018, 5:16 PM Andrew Pogrebennyk <apogrebennyk(a)sipwise.com>
wrote:
On 06/04/2018 03:39 PM, Amar Tinawi wrote:
Could this module work in scenario like in case
if the call is
established and running,
And one of the UA'a lost its connection and then reconnect and send a
new register
Could the call be continued ?
Without sending new INVITE.. just like the whatsapp
I might be wrong, but I think that GRUU and SIP-outbound RFC are
supposed to address this to some extent, because when receiving
in-dialog request the proxy can look up the new connection to the UA
using the device id and replace the connection. Not sure how this looks
like in practice and with TLS connections, since not many clients
support GRUU and some wouldn't even re-register after TCP disconnect.\
As to original question, I doubt if there is such a method for
transaction in progress, but see perhaps if topci
https://lists.cs.columbia.edu/pipermail/sip-implementors/2012-June/028480.h…
applies to you.
BR,
Andrew
2370
days inactive
2371
days old
4 comments
4 participants
participants (4)
-
Abdoul Osséni -
Alfonso Pinto -
Amar Tinawi -
Andrew Pogrebennyk