Verto: no inbound audio when verto client called from conference
Hi, I have a dialplan where a user calls a number which results in them being placed in a conference room, and an outbound called made to a verto client when the conference is created, by defining in the dialplan: <action application="conference_set_auto_outcall" data="user/1000@ $${domain}"/> The verto client (which is basically taken from the verto demo) rings and is answered. The original inbound caller can hear the verto client speaking, but the verto client can't hear the the inbound caller (although the audio from the inbound caller is getting to the conference room). If I change the dialling order - so the verto client makes the inbound call and the SIP client is dialled from the conference using the same technique as before (conference_set_auto_outcall). Then everything appears to work fine, i.e. each caller can here the other. Any suggestions as to what might be the issue or steps I could take to help identify the problem. Thanks Mark
On 4 July 2017 at 23:55, Mark Melling <mark.melling@savageminds.com> wrote: Hi,
I have a dialplan where a user calls a number which results in them being placed in a conference room, and an outbound called made to a verto client when the conference is created, by defining in the dialplan:
<action application="conference_set_auto_outcall" data="user/1000@ $${domain}"/>
The verto client (which is basically taken from the verto demo) rings and is answered.
The original inbound caller can hear the verto client speaking, but the verto client can't hear the the inbound caller (although the audio from the inbound caller is getting to the conference room).
If I change the dialling order - so the verto client makes the inbound call and the SIP client is dialled from the conference using the same technique as before (conference_set_auto_outcall). Then everything appears to work fine, i.e. each caller can here the other.
Any suggestions as to what might be the issue or steps I could take to help identify the problem.
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf -giovanni
Thanks
Mark
_________________________________________________________________________ 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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-- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
Thanks Giovanni for the suggestion. I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak). But if they dial into the conference then everything appears fine and they do hear audio. Specifically from fs_cli I entered: originate <call-url> &conference(<conf-name>@default) If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking). Whereas if you dialled into a conference from a verto client then you would hear the conference music. So I'm not sure how I can work around this. On Wed, 5 Jul 2017 at 08:57 Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On 4 July 2017 at 23:55, Mark Melling <mark.melling@savageminds.com> wrote:
Hi,
I have a dialplan where a user calls a number which results in them being placed in a conference room, and an outbound called made to a verto client when the conference is created, by defining in the dialplan:
<action application="conference_set_auto_outcall" data="user/1000@ $${domain}"/>
The verto client (which is basically taken from the verto demo) rings and is answered.
The original inbound caller can hear the verto client speaking, but the verto client can't hear the the inbound caller (although the audio from the inbound caller is getting to the conference room).
If I change the dialling order - so the verto client makes the inbound call and the SIP client is dialled from the conference using the same technique as before (conference_set_auto_outcall). Then everything appears to work fine, i.e. each caller can here the other.
Any suggestions as to what might be the issue or steps I could take to help identify the problem.
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc
Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf
-giovanni
Thanks
Mark
_________________________________________________________________________ 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
--
Sincerely,
Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ 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
On 5 July 2017 at 18:08, Mark Melling <mark.melling@savageminds.com> wrote:
Thanks Giovanni for the suggestion.
I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak).
But if they dial into the conference then everything appears fine and they do hear audio.
Specifically from fs_cli I entered:
originate <call-url> &conference(<conf-name>@default)
If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking).
Whereas if you dialled into a conference from a verto client then you would hear the conference music.
So I'm not sure how I can work around this.
Have you tried what I suggested?
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc
Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf
-giovanni
Your suggestion does work, I did the following manually from fs_cli. So with a user dialled into a conference room I was able to do:
originate <call-url> &park
Where the call-url referring to a verto client, then
uuid_transfer <verto-call-uuid> conference:<conf-name> inline
And that worked, I could hear audio from both clients connected to the conference. Thanks for your help. On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On 5 July 2017 at 18:08, Mark Melling <mark.melling@savageminds.com> wrote:
Thanks Giovanni for the suggestion.
I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak).
But if they dial into the conference then everything appears fine and they do hear audio.
Specifically from fs_cli I entered:
originate <call-url> &conference(<conf-name>@default)
If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking).
Whereas if you dialled into a conference from a verto client then you would hear the conference music.
So I'm not sure how I can work around this.
Have you tried what I suggested?
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc
Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf
-giovanni
_________________________________________________________________________ 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
Based on this I implemented a solution using lua: The key bits being: session = freeswitch.Session(<call url>) ... session:execute("sleep", 1000) session:execute("conference", <conference-name>) The only problem is that when the verto user is added to the conference they appear (in the conference) with the originating caller id and name (as set in the freeswitch.Session call), rather than their own. Is there anyway around this? Thanks Mark On Wed, 5 Jul 2017 at 22:21 Mark Melling <mark.melling@savageminds.com> wrote:
Your suggestion does work, I did the following manually from fs_cli.
So with a user dialled into a conference room I was able to do:
originate <call-url> &park
Where the call-url referring to a verto client, then
uuid_transfer <verto-call-uuid> conference:<conf-name> inline
And that worked, I could hear audio from both clients connected to the conference.
Thanks for your help.
On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On 5 July 2017 at 18:08, Mark Melling <mark.melling@savageminds.com> wrote:
Thanks Giovanni for the suggestion.
I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak).
But if they dial into the conference then everything appears fine and they do hear audio.
Specifically from fs_cli I entered:
originate <call-url> &conference(<conf-name>@default)
If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking).
Whereas if you dialled into a conference from a verto client then you would hear the conference music.
So I'm not sure how I can work around this.
Have you tried what I suggested?
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc
Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf
-giovanni
_________________________________________________________________________ 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
Set the channel caller id variables before joining the conference sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jul 7, 2017 18:35, "Mark Melling" <mark.melling@savageminds.com> wrote:
Based on this I implemented a solution using lua:
The key bits being:
session = freeswitch.Session(<call url>) ... session:execute("sleep", 1000) session:execute("conference", <conference-name>)
The only problem is that when the verto user is added to the conference they appear (in the conference) with the originating caller id and name (as set in the freeswitch.Session call), rather than their own.
Is there anyway around this?
Thanks
Mark
On Wed, 5 Jul 2017 at 22:21 Mark Melling <mark.melling@savageminds.com> wrote:
Your suggestion does work, I did the following manually from fs_cli.
So with a user dialled into a conference room I was able to do:
originate <call-url> &park
Where the call-url referring to a verto client, then
uuid_transfer <verto-call-uuid> conference:<conf-name> inline
And that worked, I could hear audio from both clients connected to the conference.
Thanks for your help.
On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
On 5 July 2017 at 18:08, Mark Melling <mark.melling@savageminds.com> wrote:
Thanks Giovanni for the suggestion.
I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak).
But if they dial into the conference then everything appears fine and they do hear audio.
Specifically from fs_cli I entered:
originate <call-url> &conference(<conf-name>@default)
If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking).
Whereas if you dialled into a conference from a verto client then you would hear the conference music.
So I'm not sure how I can work around this.
Have you tried what I suggested?
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc
Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf
-giovanni
____________________________________________________________ _____________ 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 (2)
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Giovanni Maruzzelli
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Mark Melling