ReINVITE - 488 Not Acceptable Here
Hello all, I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending. The only error I can see in the logs is: * [ERR] sofia.c:7280 Reinvite Codec Error!* The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like". Attached you can find a more detailed log file. Thank you very much in advance. Regards, -- *Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
Sergio, please fill in some additional information like which JS WebSocket client you're using, there are lots of hold issues with SipML5 and JsSIP, the only one that i know that has working hold music is sip.js (was a fork of JsSIP) and I've personally contributed towards making the hold function correctly on sip.js On Thu, Jul 23, 2015 at 5:17 AM, Sergio García <sergio.garcia@quobis.com> wrote:
Hello all,
I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending.
The only error I can see in the logs is:
* [ERR] sofia.c:7280 Reinvite Codec Error!*
The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like".
Attached you can find a more detailed log file.
Thank you very much in advance.
Regards, --
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
_________________________________________________________________________ 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 Chris, I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in the previous email, we are sending IP 0.0.0.0, port 0 and media inactive. It seems FS doesn't like this and I would like to know the reason. I've changed our end to send* a=sendonly* (instead of* a=inactive*) and to not change the port to 0, and FS accepted it. The problem now is that after one second hearing the music on hold it stops, and I can never get the media back (I'm using TURN server to avoid NAT/Firewall problems). Thank you. 2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr@gmail.com>:
Sergio, please fill in some additional information like which JS WebSocket client you're using, there are lots of hold issues with SipML5 and JsSIP, the only one that i know that has working hold music is sip.js (was a fork of JsSIP) and I've personally contributed towards making the hold function correctly on sip.js
On Thu, Jul 23, 2015 at 5:17 AM, Sergio García <sergio.garcia@quobis.com> wrote:
Hello all,
I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending.
The only error I can see in the logs is:
* [ERR] sofia.c:7280 Reinvite Codec Error!*
The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like".
Attached you can find a more detailed log file.
Thank you very much in advance.
Regards, --
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
_________________________________________________________________________ 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
-- *Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
Perhaps you could mock up a small sample app using SIP.js, check that it works properly, and then analyse the differences between what it does and what your custom SIP stack does, then? On 24 July 2015 at 04:18, Sergio García <sergio.garcia@quobis.com> wrote:
Hi Chris,
I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in the previous email, we are sending IP 0.0.0.0, port 0 and media inactive. It seems FS doesn't like this and I would like to know the reason.
I've changed our end to send* a=sendonly* (instead of* a=inactive*) and to not change the port to 0, and FS accepted it. The problem now is that after one second hearing the music on hold it stops, and I can never get the media back (I'm using TURN server to avoid NAT/Firewall problems).
Thank you.
2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr@gmail.com>:
Sergio, please fill in some additional information like which JS WebSocket client you're using, there are lots of hold issues with SipML5 and JsSIP, the only one that i know that has working hold music is sip.js (was a fork of JsSIP) and I've personally contributed towards making the hold function correctly on sip.js
On Thu, Jul 23, 2015 at 5:17 AM, Sergio García <sergio.garcia@quobis.com> wrote:
Hello all,
I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending.
The only error I can see in the logs is:
* [ERR] sofia.c:7280 Reinvite Codec Error!*
The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like".
Attached you can find a more detailed log file.
Thank you very much in advance.
Regards, --
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
_________________________________________________________________________ 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
--
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
_________________________________________________________________________ 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
why don't you check what sip.js do (sip.js is the implementation of sip in javascript better working with FS), then you model your solution behavior on it... On Fri, Jul 24, 2015 at 1:34 PM, Ben Langfeld <ben@langfeld.co.uk> wrote:
Perhaps you could mock up a small sample app using SIP.js, check that it works properly, and then analyse the differences between what it does and what your custom SIP stack does, then?
On 24 July 2015 at 04:18, Sergio García <sergio.garcia@quobis.com> wrote:
Hi Chris,
I'm not using JSSIP or Sipml5, I'm using our own solution. As I said in the previous email, we are sending IP 0.0.0.0, port 0 and media inactive. It seems FS doesn't like this and I would like to know the reason.
I've changed our end to send* a=sendonly* (instead of* a=inactive*) and to not change the port to 0, and FS accepted it. The problem now is that after one second hearing the music on hold it stops, and I can never get the media back (I'm using TURN server to avoid NAT/Firewall problems).
Thank you.
2015-07-23 18:11 GMT+02:00 Chris Tunbridge <blasterjr@gmail.com>:
Sergio, please fill in some additional information like which JS WebSocket client you're using, there are lots of hold issues with SipML5 and JsSIP, the only one that i know that has working hold music is sip.js (was a fork of JsSIP) and I've personally contributed towards making the hold function correctly on sip.js
On Thu, Jul 23, 2015 at 5:17 AM, Sergio García <sergio.garcia@quobis.com
wrote:
Hello all,
I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending.
The only error I can see in the logs is:
* [ERR] sofia.c:7280 Reinvite Codec Error!*
The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like".
Attached you can find a more detailed log file.
Thank you very much in advance.
Regards, --
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
_________________________________________________________________________ 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
--
*Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia@quobis.com | t: +34902999465
_________________________________________________________________________ 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
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
participants (4)
-
Ben Langfeld
-
Chris Tunbridge
-
Giovanni Maruzzelli
-
Sergio García