Hello, Having issue of call dropping after 32 seconds, here are the details- x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server) Outbound-proxy is set to x.x.x.174 in Gateway configuration. 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped. What is wrong here? Any help would be appreciated here. Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
5 second response: 32 seconds is a timer/[network/NAT] issue. You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call. -Avi Marcus BestFone On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies?
From where can i start? where to look?
Thanks. -- Thanks, Rutu Patel <http://www.inextrix.com> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, 1) You have very complex set-up and I doubt that you need it. 2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP. 3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty. With kind regards, Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Thanks for the reply.
Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look?
Thanks.
-- Thanks, Rutu Patel
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Jurijs 2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Thanks for the reply.
Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look?
Thanks.
-- Thanks, Rutu Patel
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Jurijs, We have to consider the setup with freeswitch and opensips only. registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) And there is a firewall and we have added x.x.x.3 IP there. Is it possible there is some network related issue Or any sip profile parameter we need to set? -- Thanks, Rutu Patel On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo@gmail.com> wrote:
Hi,
1) You have very complex set-up and I doubt that you need it.
2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP.
3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty.
With kind regards,
Jurijs
2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Thanks for the reply.
Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look?
Thanks.
-- Thanks, Rutu Patel
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Jurijs
2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Thanks for the reply.
Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look?
Thanks.
-- Thanks, Rutu Patel
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Rutu, Firewall should not be an issue. If you think that it may change SIP packets, you can always use TLS. I doubt that it is good idea to add one more server in set-up, just because of firewall, you just need to configure all of your servers, devices properly. If you can, you should eliminate unnecessary servers from your set-up.
From what you described before, it might be issue not connected to NAT, but because Opensips wasn't configured properly. I had similar issue when Kamailio(Opensip is fork from Kamailio project, so they almost identical) was wrongly configured, particularly path header wasn't inserted by Kamailio. But this 30 seconds timeout is quite often NAT issue, but again if you have NAT issue, you should not blame FW or anything else, you should just configure your Freeswitch and Opensips properly. Almost all devices in internet are behind NAT and almost all of them works perfectly with VoIP.
So if you need, help, then please send full sip trace, so I can take a look on it. With kind regards, Jurijs 2016-02-29 15:15 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Hi Jurijs,
We have to consider the setup with freeswitch and opensips only. registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server)
And there is a firewall and we have added x.x.x.3 IP there.
Is it possible there is some network related issue Or any sip profile parameter we need to set?
-- Thanks, Rutu Patel
On Tue, Feb 23, 2016 at 1:07 PM, Jurijs Ivolga <jurij.ivo@gmail.com> wrote:
Hi,
1) You have very complex set-up and I doubt that you need it.
2) As far as you have user with ip x.x.x.174 and opensips server with same ip x.x.x.174 it very hard to debug. So I propose you to send new log where will be difference between user ip and opensips IP.
3) If you have possibility, try to register directly with a user to x.x.x.3 gateway and check if same issue still exists, if there is no such issue anymore, then thee is definitely issue in your opensips x.x.x.174 and freeswitch x.x.x.166. My point here is that you need to isolate issue and to understand what part of your set-up works as expected and what is faulty.
With kind regards,
Jurijs
2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Thanks for the reply.
Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look?
Thanks.
-- Thanks, Rutu Patel
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Jurijs
2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel@inextrix.com>:
Thanks for the reply.
Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies? From where can i start? where to look?
Thanks.
-- Thanks, Rutu Patel
On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Avi Marcus
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Jurijs Ivolga
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Rutu Patel