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
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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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@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@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@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.ht... applies to you.
BR, Andrew
Thanks Andrew Even if i don't Use tls ?
On Mon, Jun 4, 2018, 5:16 PM Andrew Pogrebennyk apogrebennyk@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.ht... applies to you.
BR, Andrew
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Abdoul Osséni -
Alfonso Pinto -
Amar Tinawi -
Andrew Pogrebennyk