Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite
Hello, This is my first posting into this list although I monitored activities in the past. Freeswitch behaves great in our environment - thank you all who contribute! We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please: We have incoming calls from provider to FS that further bridge to different Standard SIP PBX. Call is established to SIP device connected on the PBX. The PBX immediately invites back attempting to redirect rtp path to start recording the call. FS starts sending RTP to the new destination. There are 2 different very similar PBX environments connected to the same FS: - In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds. - In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path. The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values <!-- rtp inactivity timeout --> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> Are there any other parameters that could have such an effect? Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated! Thank you very much! Regards
Hello again, The following log message coincides with the initiation of the issue: 2017-02-13 12:19:52.549786 [INFO] switch_rtp.c:5867 Auto Changing port from 172.17.85.240:58638 to 172.17.85.240:58632 Version shows: FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) Tried adding in the default dialplan, under the inbound extension, just before the bridge command: <action application="set" data="disable_rtp_auto_adjust=true"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> Where is the correct place to add them? Are these parameters valid for my exact version? How can I verify if they become active? Adding a line like the one below shows blank. <action application="log" data="WARNING ~ Inbound disable-rtp-auto-adjust value:${disable-rtp-auto-adjust}"/> Thank you From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bejan, Alex Sent: Monday, January 30, 2017 4:12 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite Hello, This is my first posting into this list although I monitored activities in the past. Freeswitch behaves great in our environment - thank you all who contribute! We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please: We have incoming calls from provider to FS that further bridge to different Standard SIP PBX. Call is established to SIP device connected on the PBX. The PBX immediately invites back attempting to redirect rtp path to start recording the call. FS starts sending RTP to the new destination. There are 2 different very similar PBX environments connected to the same FS: - In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds. - In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path. The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values <!-- rtp inactivity timeout --> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> Are there any other parameters that could have such an effect? Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated! Thank you very much! Regards
1.4.x has been EOL for some time, you should try the latest 1.6.x release too! ./b On Tue, Feb 14, 2017 at 9:01 AM Bejan, Alex <alex.bejan@necam.com> wrote:
Hello again,
The following log message coincides with the initiation of the issue:
2017-02-13 12:19:52.549786 [INFO] switch_rtp.c:5867 Auto Changing port from 172.17.85.240:58638 to 172.17.85.240:58632
Version shows: FreeSWITCH Version 1.4.26-37~64bit (-37 64bit)
Tried adding in the default dialplan, under the inbound extension, just before the bridge command:
<action application="set" data="disable_rtp_auto_adjust=true"/>
<action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
Where is the correct place to add them?
Are these parameters valid for my exact version?
How can I verify if they become active? Adding a line like the one below shows blank.
<action application="log" data="WARNING ~ Inbound disable-rtp-auto-adjust value:${disable-rtp-auto-adjust}"/>
Thank you
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Bejan, Alex *Sent:* Monday, January 30, 2017 4:12 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite
Hello,
This is my first posting into this list although I monitored activities in the past.
Freeswitch behaves great in our environment – thank you all who contribute!
We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please:
We have incoming calls from provider to FS that further bridge to different Standard SIP PBX.
Call is established to SIP device connected on the PBX.
The PBX immediately invites back attempting to redirect rtp path to start recording the call.
FS starts sending RTP to the new destination.
There are 2 different very similar PBX environments connected to the same FS:
- In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds.
- In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path.
The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values
<!-- rtp inactivity timeout -->
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
Are there any other parameters that could have such an effect?
Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated!
Thank you very much!
Regards
_________________________________________________________________________ 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
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch 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) *Skype:*briankwest
Thank you very much for the answer Brian! We had a system that was working fine and it was on version 1.4.26. Tried using 1.6 and ran into a different issue where the MOH provided by the Standard SIP PBX would not make it out to the trunk (FS MOH was played instead). So we had to revert back (temporary) to the known working version, with intent to upgrade later. We will try again version 1.6. Will this allow disabling the rtp_auto_adjust? Is there any difference in the time before it takes effect? Is there any way we could either disable this behavior or increase the time until it takes effect in 1.4.26? Regards, Alex Bejan Brian West brian at freeswitch.org <mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%5D%20one%20way%20speech%20path%20after%20attempting%20voice%0A%20path%20redirecting%20in%20new%20invite&In-Reply-To=58e9aed459074db1baacf5893ce1803c%40DLNCACITEXMBX03.na-idm.na-gad.nec.com> Wed Feb 15 15:19:46 MSK 2017 1.4.x has been EOL for some time, you should try the latest 1.6.x release too! ./b
There is an option in 1.6 to proxy the hold across, but you should be testing in a lab before going into production. On Wed, Feb 15, 2017 at 12:17 PM, Bejan, Alex <alex.bejan@necam.com> wrote:
Thank you very much for the answer Brian!
We had a system that was working fine and it was on version 1.4.26. Tried using 1.6 and ran into a different issue where the MOH provided by the Standard SIP PBX would not make it out to the trunk (FS MOH was played instead). So we had to revert back (temporary) to the known working version, with intent to upgrade later.
We will try again version 1.6. Will this allow disabling the rtp_auto_adjust? Is there any difference in the time before it takes effect?
Is there any way we could either disable this behavior or increase the time until it takes effect in 1.4.26?
Regards,
Alex Bejan
*Brian West* brian at freeswitch.org <freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%5D%20one%20way%20speech%20path%20after%20attempting%20voice%0A%20path%20redirecting%20in%20new%20invite&In-Reply-To=58e9aed459074db1baacf5893ce1803c%40DLNCACITEXMBX03.na-idm.na-gad.nec.com> *Wed Feb 15 15:19:46 MSK 2017*
1.4.x has been EOL for some time, you should try the latest 1.6.x release too!
./b
_________________________________________________________________________ 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 Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch 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) *Skype:*briankwest
participants (2)
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Bejan, Alex
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Brian West