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Do you have to use Portsip ? FS does webrtc.<br>
Anyway, have you tried media_webrtc=false ? <br>
<br>
<a class="moz-txt-link-freetext" href="https://wiki.freeswitch.org/wiki/Variable_media_webrtc">https://wiki.freeswitch.org/wiki/Variable_media_webrtc</a><br>
<br>
I would try it by :<br>
<action application="bridge" data="{media_webrtc=false}sofia/....<br>
<br>
but I'm not sure if it works this way, I've never been trying this
before.<br>
<br>
Jan<br>
<br>
<br>
<br>
<br>
<br>
<div class="moz-cite-prefix">On 23.08.2016 11:00, james wrote:<br>
</div>
<blockquote cite="mid:tencent_313189E80099985F1BD146BD@qq.com"
type="cite">
<div>Hello:</div>
<div>I am using portsip webRTC gateway to test the compatibility.
I use two phones for webRTC testing. One is normal webRTC client
and another one is softphone.</div>
<div>FreeSWITCH work as a IPPBX. <span style="line-height: 1.5;">The
call flows are:</span></div>
<div><span style="color: rgb(204, 0, 0); font-family: Arial,
Helvetica; font-size: 13px; line-height: 18px;"></span></div>
<div><img src="cid:part1.F1E22F0B.70F6EC1A@vmtele.com"> </div>
<div>the problem is that webRTC calls sip phone without any
problem, but sip phone calls to webRTC than failed. </div>
<div>errors are:</div>
<div>
<div>2016-08-23 16:29:39.971775 [DEBUG]
switch_core_state_machine.c:569
(<a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid">sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid</a>) State Change
CS_REPORTING -> CS_DESTROY</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.971775 [DEBUG]
switch_core_session.c:1647 Session 171
(<a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid">sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid</a>) Locked, Waiting
on external entities</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [DEBUG]
switch_ivr_originate.c:3759 Originate Resulted in Error Cause:
88 [INCOMPATIBLE_DESTINATION]</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [NOTICE]
switch_ivr_originate.c:2771 Cannot create outgoing channel of
type [user] cause: [INCOMPATIBLE_DESTINATION]</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [DEBUG]
switch_ivr_originate.c:3759 Originate Resulted in Error Cause:
88 [INCOMPATIBLE_DESTINATION]</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401
Originate Failed. Cause: INCOMPATIBLE_DESTINATION</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>EXECUTE <a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/1004@120.26.130.142">sofia/internal/1004@120.26.130.142</a> answer()</div>
</div>
<div>-------------------------</div>
<div>It looks freeswitch send with ICE to webRTC. Does anyone know
how i do disable the webRTC negotiation and only send normal sip
info to webRTC gateway<span style="color: rgb(118, 118, 118);
font-family: Arial, Helvetica; font-size: 13px; line-height:
20px;"></span></div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
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