Re: [Freeswitch-users] Handle media of all calls via port 443?
I did: fs_cli -x 'fsctl shutdown elegant restart' but that still yields only the same candidate as before. Then I rebooted the vm (ec2), tried again, and there was still just one candidate of the same type as before. On Wed, Aug 12, 2020 at 5:18 AM < freeswitch-users-request@lists.freeswitch.org> wrote:
---------- Forwarded message ---------- From: Giovanni Maruzzelli <gmaruzz@gmail.com>
In this case, you *must* restart freeswitch, reloadxml is not enough -giovanni
On Tue, Aug 11, 2020 at 9:01 PM David P <davidswalkabout@gmail.com> wrote:
Thank you for your suggestion, Sergey. However, I found that introducing those two settings (in FSv10.4 on Debian10) made no difference; I still got just one candidate in FS's answer SDP and it's udp rather than tcp (for port 443)...
Before introducing <X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/> in /etc/freeswitch/vars.xml and <action application="set" data="media_webrtc=true"/> in /etc/freeswitch/dialplan/mydialplan.xml the browser receives Answer SDP containing only one ice candidate: a=candidate:5500513041 1 udp 659136 35.xxx.yy.zzz 26928 typ host generation 0
After introducing those two config changes, and `fs_cli -x "reloadxml"`, the set of candidates in the answer the browser gets is the same: a=candidate:4793838179 1 udp 659136 35.xxx.yy.zzz 19518 typ host generation 0
The reason I asked about supporting all media (and signaling) via port 443 is that some users are in firewalls that block all ports except 443 (or 80).
On Mon, Aug 10, 2020 at 5:14 AM < freeswitch-users-request@lists.freeswitch.org> wrote:
---------- Forwarded message ---------- From: Sergey Safarov <s.safarov@gmail.com>
You can offer ice for all calls from FreeSwitch without coturn
<X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/>
More details
https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal#NATTraver...
also, you can try the variable "media_webrtc=true"
On Sun, Aug 9, 2020 at 8:42 PM David P <davidswalkabout@gmail.com> wrote:
Can FS v1.10 be configured to handle SRTP of all calls via port 443?
This would allow us to stop using coturn, which is often too slow to provide a TLS relay candidate before browsers finish gathering ice candidates.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Sincerely,
Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
---------- Forwarded message ---------- From: Prashant Lamba <prashant.lamba@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Wed, 12 Aug 2020 13:48:26 +0530 Subject: Re: [Freeswitch-users] Change name shown on the ITSP or registered users You can do it in the SIP profile (under "user-agent-string"). But why would you want to hide the fact that you're using FS :-)))
Prashant -- To save our tigers, save their habitat. Think before you print this email. <www.saveourtigers.com>
On Tue, Aug 11, 2020 at 1:49 AM John Tuxies <atuxnull@gmail.com> wrote:
I would like to change the Freeswitch name shown when a users connects to or to the ITSP. So they will not know the name and version of FS. I would like to name it something like sip.lab.mydomain.org
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
On Wed, Aug 12, 2020, 20:22 David P <davidswalkabout@gmail.com> wrote:
I did: fs_cli -x 'fsctl shutdown elegant restart' but that still yields only the same candidate as before. Then I rebooted the vm (ec2), tried again, and there was still just one candidate of the same type as before.
So, that was not the solution, but at least you actually tested it!
On Wed, Aug 12, 2020 at 5:18 AM < freeswitch-users-request@lists.freeswitch.org> wrote:
---------- Forwarded message ---------- From: Giovanni Maruzzelli <gmaruzz@gmail.com>
In this case, you *must* restart freeswitch, reloadxml is not enough -giovanni
On Tue, Aug 11, 2020 at 9:01 PM David P <davidswalkabout@gmail.com> wrote:
Thank you for your suggestion, Sergey. However, I found that introducing those two settings (in FSv10.4 on Debian10) made no difference; I still got just one candidate in FS's answer SDP and it's udp rather than tcp (for port 443)...
Before introducing <X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/> in /etc/freeswitch/vars.xml and <action application="set" data="media_webrtc=true"/> in /etc/freeswitch/dialplan/mydialplan.xml the browser receives Answer SDP containing only one ice candidate: a=candidate:5500513041 1 udp 659136 35.xxx.yy.zzz 26928 typ host generation 0
After introducing those two config changes, and `fs_cli -x "reloadxml"`, the set of candidates in the answer the browser gets is the same: a=candidate:4793838179 1 udp 659136 35.xxx.yy.zzz 19518 typ host generation 0
The reason I asked about supporting all media (and signaling) via port 443 is that some users are in firewalls that block all ports except 443 (or 80).
On Mon, Aug 10, 2020 at 5:14 AM < freeswitch-users-request@lists.freeswitch.org> wrote:
---------- Forwarded message ---------- From: Sergey Safarov <s.safarov@gmail.com>
You can offer ice for all calls from FreeSwitch without coturn
<X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/>
More details
https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal#NATTraver...
also, you can try the variable "media_webrtc=true"
On Sun, Aug 9, 2020 at 8:42 PM David P <davidswalkabout@gmail.com> wrote:
Can FS v1.10 be configured to handle SRTP of all calls via port 443?
This would allow us to stop using coturn, which is often too slow to provide a TLS relay candidate before browsers finish gathering ice candidates.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Sincerely,
Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
---------- Forwarded message ---------- From: Prashant Lamba <prashant.lamba@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Wed, 12 Aug 2020 13:48:26 +0530 Subject: Re: [Freeswitch-users] Change name shown on the ITSP or registered users You can do it in the SIP profile (under "user-agent-string"). But why would you want to hide the fact that you're using FS :-)))
Prashant -- To save our tigers, save their habitat. Think before you print this email. <www.saveourtigers.com>
On Tue, Aug 11, 2020 at 1:49 AM John Tuxies <atuxnull@gmail.com> wrote:
I would like to change the Freeswitch name shown when a users connects to or to the ITSP. So they will not know the name and version of FS. I would like to name it something like sip.lab.mydomain.org
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Hi Brian W, Back on Aug 10, Sergey Safarov suggested we might get FS's answer sdp to include more candidates via variable "media_webrtc=true". (We'd like TCP via port 443.) The only documentation I can find is an old post by you: http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727... It didn't work for me when I set "media_webrtc" as a regular dialplan var, but your post mentions the {} of a bridge. We use a conference, so should we set it this way? <action application="conference_set_auto_outcall" data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_webrtc=true']sofia/gateway/{hidden}"/> On Thu, Aug 13, 2020 at 6:21 AM David P <davidswalkabout@gmail.com> wrote:
I did: fs_cli -x 'fsctl shutdown elegant restart' but that still yields only the same candidate as before. Then I rebooted the vm (ec2), tried again, and there was still just one candidate of the same type as before.
On Wed, Aug 12, 2020 at 5:18 AM < freeswitch-users-request@lists.freeswitch.org> wrote:
---------- Forwarded message ---------- From: Giovanni Maruzzelli <gmaruzz@gmail.com>
In this case, you *must* restart freeswitch, reloadxml is not enough -giovanni
On Tue, Aug 11, 2020 at 9:01 PM David P <davidswalkabout@gmail.com> wrote:
Thank you for your suggestion, Sergey. However, I found that introducing those two settings (in FSv10.4 on Debian10) made no difference; I still got just one candidate in FS's answer SDP and it's udp rather than tcp (for port 443)...
Before introducing <X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/> in /etc/freeswitch/vars.xml and <action application="set" data="media_webrtc=true"/> in /etc/freeswitch/dialplan/mydialplan.xml the browser receives Answer SDP containing only one ice candidate: a=candidate:5500513041 1 udp 659136 35.xxx.yy.zzz 26928 typ host generation 0
After introducing those two config changes, and `fs_cli -x "reloadxml"`, the set of candidates in the answer the browser gets is the same: a=candidate:4793838179 1 udp 659136 35.xxx.yy.zzz 19518 typ host generation 0
The reason I asked about supporting all media (and signaling) via port 443 is that some users are in firewalls that block all ports except 443 (or 80).
On Mon, Aug 10, 2020 at 5:14 AM < freeswitch-users-request@lists.freeswitch.org> wrote:
---------- Forwarded message ---------- From: Sergey Safarov <s.safarov@gmail.com>
You can offer ice for all calls from FreeSwitch without coturn
<X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/>
More details
https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal#NATTraver...
also, you can try the variable "media_webrtc=true"
On Sun, Aug 9, 2020 at 8:42 PM David P <davidswalkabout@gmail.com> wrote:
Can FS v1.10 be configured to handle SRTP of all calls via port 443?
This would allow us to stop using coturn, which is often too slow to provide a TLS relay candidate before browsers finish gathering ice candidates.
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
-- Sincerely,
Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
---------- Forwarded message ---------- From: Prashant Lamba <prashant.lamba@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Wed, 12 Aug 2020 13:48:26 +0530 Subject: Re: [Freeswitch-users] Change name shown on the ITSP or registered users You can do it in the SIP profile (under "user-agent-string"). But why would you want to hide the fact that you're using FS :-)))
Prashant -- To save our tigers, save their habitat. Think before you print this email. <www.saveourtigers.com>
On Tue, Aug 11, 2020 at 1:49 AM John Tuxies <atuxnull@gmail.com> wrote:
I would like to change the Freeswitch name shown when a users connects to or to the ITSP. So they will not know the name and version of FS. I would like to name it something like sip.lab.mydomain.org
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
FS won’t but a client that supports turn could use that via the turn server.
On Sep 3, 2020, at 10:07 PM, David P <davidswalkabout@gmail.com> wrote:
Hi Brian W,
Back on Aug 10, Sergey Safarov suggested we might get FS's answer sdp to include more candidates via variable "media_webrtc=true". (We'd like TCP via port 443.) The only documentation I can find is an old post by you: http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727... <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html>
It didn't work for me when I set "media_webrtc" as a regular dialplan var, but your post mentions the {} of a bridge. We use a conference, so should we set it this way?
<action application="conference_set_auto_outcall" data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_webrtc=true']sofia/gateway/{hidden}"/>
participants (3)
-
David P
-
Giovanni Maruzzelli
-
Mike Jerris