On Mon, Aug 20, 2012 at 11:53 AM, Joshua Colp <jc...@digium.com> wrote: > ----- Original Message ----- >> On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro <jcas...@instant.com.br> >> wrote: >> > On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro >> > <jcas...@instant.com.br> wrote: >> >> I still get "unauthorized" from sipml5 with these modifications. I >> >> used port 80 instead of 8088 (no other webserver listening on 80), >> >> was >> >> that wrong? >> > >> > Correction. It's actually "Failed to connect to the server". I set >> > the >> > proxy address and port correctly in sipml5's call.htm (it registers >> > on >> > Kamailio). >> >> ...which is in fact a 404 response from Asterisk. Here's the response >> I received: http://users.vialink.com.br/jcastro/refused.cap >> >> I suspect I am configuring something wrong, but what is it? > > The complete URL to use is http://<asterisk IP address or host>:8088/ws > > Note the /ws at the end. WebSocket support is only available there. Doing > otherwise would have required core HTTP server changes, which I wanted to > avoid. Depending on what you are testing with you may need to change it > slightly to add that in.
Well, I did the following changes in sipml5 and now I get a "Bad Request" on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: Index: call.htm =================================================================== --- call.htm (revision 68) +++ call.htm (working copy) @@ -351,8 +351,9 @@ // we will connect to one of them and let the balancer to choose the right one (less connected sockets) // each port can accept up to 65K connections which means that the cloud can manage 325K active connections // the number of port will be increased or decreased based on the current trafic - i_port = 4062 + (((new Date().getTime()) % 5) * 1000); - s_proxy = "sipml5.org"; + // i_port = 4062 + (((new Date().getTime()) % 5) * 1000); + i_port = 80; + s_proxy = "192.168.0.111"; } // create a new SIP stack. Not mandatory as it's possible to reuse the same satck Index: src/tinySIP/src/tsip_stack.js =================================================================== --- src/tinySIP/src/tsip_stack.js (revision 68) +++ src/tinySIP/src/tsip_stack.js (working copy) @@ -351,7 +351,7 @@ return -2; } - tsk_utils_log_info("SIP stack start: proxy='" + this.network.s_proxy_cscf_host + ":" + this.network.i_proxy_cscf_port + "', realm='" + this.network.o_uri_realm + "', impi='" + this.identity.s_impi + "', impu='" + this.identity.o_uri_impu + "'"); + tsk_utils_log_info("SIP stack start: proxy='" + this.network.s_proxy_cscf_host + ":" + this.network.i_proxy_cscf_port + "/ws', realm='" + this.network.o_uri_realm + "', impi='" + this.identity.s_impi + "', impu='" + this.identity.o_uri_impu + "'"); this.network.o_transport = this.o_layer_transport.transport_new(this.network.e_proxy_cscf_type, this.network.s_proxy_cscf_host, this.network.i_proxy_cscf_port, "SIP Transport", __tsip_stack_transport_callback); if (!this.network.o_transport) { @@ -716,4 +716,4 @@ } return 0; -} \ No newline at end of file +} Index: src/tinySIP/src/transports/tsip_transport.js =================================================================== --- src/tinySIP/src/transports/tsip_transport.js (revision 68) +++ src/tinySIP/src/transports/tsip_transport.js (working copy) @@ -368,7 +368,7 @@ return -1; } - var s_url = tsk_string_format("{0}://{1}:{2}",o_self.s_protocol, o_self.s_host, o_self.i_port); + var s_url = tsk_string_format("{0}://{1}:{2}/ws",o_self.s_protocol, o_self.s_host, o_self.i_port); tsk_utils_log_info("Connecting to '"+s_url+"'"); o_self.o_ws = new WebSocket(s_url, 'sip'); o_self.o_ws.binaryType = "arraybuffer"; @@ -458,7 +458,7 @@ } var b_isInternetExplorer = (WebRtc4all_GetType() == WebRtcType_e.IE); - var s_url = tsk_string_format("{0}://{1}:{2}",o_self.s_protocol, o_self.s_host, o_self.i_port); + var s_url = tsk_string_format("{0}://{1}:{2}/ws",o_self.s_protocol, o_self.s_host, o_self.i_port); tsk_utils_log_info("Connecting to '"+s_url+"'"); if(b_isInternetExplorer){ o_self.o_transport = new ActiveXObject("webrtc4ie.NetTransport"); @@ -480,7 +480,7 @@ if(o_self.o_transport.defaultDestAddr && o_self.o_transport.defaultDestPort){ o_self.s_host = o_self.o_transport.defaultDestAddr; o_self.i_port = o_self.o_transport.defaultDestPort; - tsk_utils_log_info("Transport default destination=" + o_self.s_host + ":" + o_self.i_port); + tsk_utils_log_info("Transport default destination=" + o_self.s_host + ":" + o_self.i_port + "/ws"); } o_self.b_started = true; o_self.signal(tsip_transport_event_type_e.STARTED, "Network transport started", null); Index: src/tinyMEDIA/src/tmedia_session_jsep.js =================================================================== --- src/tinyMEDIA/src/tmedia_session_jsep.js (revision 68) +++ src/tinyMEDIA/src/tmedia_session_jsep.js (working copy) @@ -168,6 +168,8 @@ tmedia_session_jsep.prototype.decorate_lo = function () { if (this.o_sdp_lo) { + this.o_sdp_lo.remove_media("video"); + /* Session name for debugging */ var o_hdr_S; if ((o_hdr_S = this.o_sdp_lo.get_header(tsdp_header_type_e.S))) { @@ -278,4 +280,4 @@ this.o_sdp_ro = null; return 0; -} \ No newline at end of file +} -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users