Freeswitch behind NAT with 2 external providers: No audio
Hello, I have the following scenario * Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip * Echo from an external SIP provider test inbound works with audio * Calling outbound to the same external SIP provider works with audio * *Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio* * Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio I also set up another (modified NAT) profile * like external profile - but with a different ext_rtp_ip * calling outbound to via NAT profile to another SIP provider works with audio * *Receiving a call from the external SIP provider (external profile) and bridging it to another number ****at the another external SIP provider ***(nat***profile)** does not have audio* * Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider* * So I think, there is a problem with briding a natted external call to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine. Btw. Freeswitch logs do show RTCPpacket not written right after bridginbg the call. Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Its going to be because you probably need to pre_answer, play a little media before you hair pin the call back out the same provider. :) playback silence_stream://1000 after pre_answer would probably fix it prior to bridge. On Wed, Mar 9, 2016 at 8:32 AM, Peter Steinbach <lists@telefaks.de> wrote:
Hello,
I have the following scenario
- Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip - Echo from an external SIP provider test inbound works with audio - Calling outbound to the same external SIP provider works with audio - *Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio* - Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio
I also set up another (modified NAT) profile
- like external profile - but with a different ext_rtp_ip - calling outbound to via NAT profile to another SIP provider works with audio - *Receiving a call from the external SIP provider (external profile) and bridging it to another number **at the another external SIP provider **(nat profile) does not have audio* - Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider
So I think, there is a problem with briding a natted external call to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine. Btw. Freeswitch logs do show RTCPpacket not written right after bridginbg the call.
Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help?
-- With kind regards Peter Steinbach
Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de Internet: www.telefaks.de
_________________________________________________________________________ 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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Herr Brian, thank you, that did the trick! I added a pre_aswer, a playback and finally a ring_ready in order to playback a ring tone to the caller. Now we have 2 way audio. Best regards Peter On 03/09/16 15:47, Brian West wrote:
Its going to be because you probably need to pre_answer, play a little media before you hair pin the call back out the same provider. :) playback silence_stream://1000 after pre_answer would probably fix it prior to bridge.
On Wed, Mar 9, 2016 at 8:32 AM, Peter Steinbach <lists@telefaks.de <mailto:lists@telefaks.de>> wrote:
Hello,
I have the following scenario
* Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip * Echo from an external SIP provider test inbound works with audio * Calling outbound to the same external SIP provider works with audio * *Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio* * Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio
I also set up another (modified NAT) profile
* like external profile - but with a different ext_rtp_ip * calling outbound to via NAT profile to another SIP provider works with audio * *Receiving a call from the external SIP provider (external profile) and bridging it to another number **at the another external SIP provider ***(nat***profile)** does not have audio* * Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider* *
So I think, there is a problem with briding a natted external call to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine. Btw. Freeswitch logs do show RTCPpacket not written right after bridginbg the call.
Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help?
-- With kind regards Peter Steinbach
Telefaks Services GmbH mailto:lists (att) telefaks.de <http://telefaks.de> Internet: www.telefaks.de <http://www.telefaks.de>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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
--
*/Brian West/* brian@freeswitch.org <mailto:brian@freeswitch.org>
*/Twitter: @FreeSWITCH , @briankwest/* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
participants (2)
-
Brian West
-
Peter Steinbach