JsSip Client receiving "488 Not Acceptable Here" when trying to make a call
Hi Everyone, I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here. The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found. I'v attached the call trace, appreciate any help. Thanks, Mahmood Alkhalil.
Hi Mahmood, Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip. var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, }; var session = ua.call(uri, options); kind regards, Piotr Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil < mahmood.alkhalil@outlook.com> wrote:
Hi Everyone,
I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here.
The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found.
I'v attached the call trace, appreciate any help.
Thanks, Mahmood Alkhalil. _________________________________________________________________________
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
Get a trace and check your codecs. On Mon, 17 Feb 2020 at 22:57, Piotr Gregor <piotr@dataandsignal.com> wrote:
Hi Mahmood,
Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip.
var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, };
var session = ua.call(uri, options);
kind regards, Piotr
Piotr Gregor Software Engineer
M: (+44) 07483 866 525 www: dataandsignal.com
On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil < mahmood.alkhalil@outlook.com> wrote:
Hi Everyone,
I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here.
The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found.
I'v attached the call trace, appreciate any help.
Thanks, Mahmood Alkhalil. _________________________________________________________________________
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
-- Regards, David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
Hi Piotr/David, Thanks for you help. turns out JsSip send wrong SDP "m=" field, I've started using SipJS library and it's working fine without any issues. i was wondering if anyone is using SipJS (https://sipjs.com/) in production environment and can provide a feedback on its stability and quality, I'm planning to start building a project using SipJS library, so would really appreciate any feedback On 2/18/20 5:51 AM, David Villasmil wrote: Get a trace and check your codecs. On Mon, 17 Feb 2020 at 22:57, Piotr Gregor <piotr@dataandsignal.com<mailto:piotr@dataandsignal.com>> wrote: Hi Mahmood, Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip. var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, }; var session = ua.call(uri, options); kind regards, Piotr [https://drive.google.com/a/dataandsignal.com/uc?id=1lUdzdLjEycxXcGhDz-bKqDX8...] Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com<http://dataandsignal.com> On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil <mahmood.alkhalil@outlook.com<mailto:mahmood.alkhalil@outlook.com>> wrote: Hi Everyone, I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here. The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found. I'v attached the call trace, appreciate any help. Thanks, Mahmood Alkhalil. _________________________________________________________________________ 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<mailto: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<mailto: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<mailto: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<mailto: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 -- Regards, David Villasmil email: david.villasmil.work@gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 _________________________________________________________________________ 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<mailto: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<mailto: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
SIP.js is good and well supported On Fri, Feb 28, 2020 at 5:09 PM Mahmood Alkhalil < mahmood.alkhalil@outlook.com> wrote:
Hi Piotr/David,
Thanks for you help.
turns out JsSip send wrong SDP "m=" field, I've started using SipJS library and it's working fine without any issues.
i was wondering if anyone is using SipJS (https://sipjs.com/) in production environment and can provide a feedback on its stability and quality, I'm planning to start building a project using SipJS library, so would really appreciate any feedback On 2/18/20 5:51 AM, David Villasmil wrote:
Get a trace and check your codecs.
On Mon, 17 Feb 2020 at 22:57, Piotr Gregor <piotr@dataandsignal.com> wrote:
Hi Mahmood,
Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip.
var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, };
var session = ua.call(uri, options);
kind regards, Piotr
Piotr Gregor Software Engineer
M: (+44) 07483 866 525 www: dataandsignal.com
On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil < mahmood.alkhalil@outlook.com> wrote:
Hi Everyone,
I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here.
The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found.
I'v attached the call trace, appreciate any help.
Thanks, Mahmood Alkhalil. _________________________________________________________________________
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
-- Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
_________________________________________________________________________
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 Servicessales@freeswitch.comhttps://freeswitch.com
Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.c...
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswit...
_________________________________________________________________________
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
participants (4)
-
David Villasmil
-
Giovanni Maruzzelli
-
Mahmood Alkhalil
-
Piotr Gregor