Re: [asterisk-users] PJSIP - Video Support for WebRTC
Matthew Jordan digium.com> writes: > > On Mon, Mar 23, 2015 at 8:55 AM, Gosmac gmail.com> wrote: > > Hey i have an interesting topic to discuss here. > > > > The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 > it is a feature that definitely asterisk 13 should support . > > > > the problems that i faced with this is the following and i hope i could get an advise here. > > > > asterisk 13 vanilla version has some issues marking the video packets this complain web browser > specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking > video streams :) it just mark video packets not touch anything else and web browser show video on web page > now I’m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try > echo() dialplan test and everything work perfect on echo test :). > > > > i have two questions and i hope you could give me some advise. > > > > 1) after marking video packet I’m able to make Dial() between two webrtc peers but i get one way audio and > video on callee party, “after 3 minutes on call” i get two way audio and video on all parties seems to be > not just a problem on a missing keyframe. > > > > 1.1) the 3 minutes delay only happen using chrome stable , could be a dtls problem when asterisk make an > offer to other endpoint? > > 1.2) when i use chrome-dev and i disable dlts encryption everything work perfect on video call. > > > > 2) after marking video packets i realize that when you make a call with video and you involve on dialplan an > application like playback or music on hold any application that played audio files (audio and video never work). > > > > 2.1) asterisk is muggling the audio and video streams ? > > > > This is good information for all guys out there that wants to support video on webrtc in asterisk 13 > > > > Please stop spamming the list with this e-mail. Resending it multiple > times is clearly not yielding the results you'd like. > Hi Matthew, I'm testing WebRTC (JSSIP) with Asterisk 12.8 after following the https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support link. Using Firefox, I can connect both JSSIP Clients to asterisk. When I Call one Client, the Client just Ring One Time and after pick up a receive WebRTC error on the Firefox browser. Here is my asterisk sip debug: <--- SIP read from WS:192.168.2.103:49851 ---> INVITE sip:6000@192.168.2.106 SIP/2.0 Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689 Max-Forwards: 69 To: From: "6001" ;tag=m6bqn333dr Call-ID: 1ansppdrpdulbtr3j5ub CSeq: 6407 INVITE X-Can-Renegotiate: false Contact: Content-Type: application/sdp Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: ice,replaces,outbound User-Agent: JsSIP 2.0.2 Content-Length: 3158 v=0 o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0 s=- t=0 0 a=sendrecv a=fingerprint:sha-256 E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF: C1:23:72:03:F6:61:CC:F6 a=group:BUNDLE sdparta_0 sdparta_1 a=ice-options:trickle a=msid-semantic:WMS * m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102 a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a 56806 typ host a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa 56807 typ host a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0 2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a 56810 typ host a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa 56811 typ host a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr 192.168.2.103 rport 56808 a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr 192.168.2.103 rport 56812 a=sendrecv a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice- pwd:138f583004cb3079134e8e8f20dac36f a=ice-ufrag:0941ac54 a=mid:sdparta_0 a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe} {bba6da45-42c8-4529-8f4b-046cffcdc40d} a=rtcp:56812 IN IP4 87.169.189.102 a=rtcp-mux a=rtpmap:109 opus/48000/2 a=rtpmap:9 G722/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=setup:actpass a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c} m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102 a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a 56814 typ host a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa 56815 typ host a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0 2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a 60291 typ host a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc
[asterisk-users] SIP trunk
It seems I am not getting any digits coming over a SIP trunk. How can I match "anything" or "nothing" and start my extension. Usually I have something like: exten => 55,1,Goto(,yyy,1) but if 55 does not come across and it appears to be no digits coming across how do I match that that and just start. I thought about _X but that says digits. I dont think I am getting any digits I just want *anything* coming across to start the call. Basically ANY call coming across the trunk just do the same as 55 above. Thanks, Jerry -- _ -- 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
Re: [asterisk-users] SIP trunk
_. ? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Tue, Jul 26, 2016 at 11:39 AM, Jerry Geis wrote: > It seems I am not getting any digits coming over a SIP trunk. > > How can I match "anything" or "nothing" and start my extension. > > Usually I have something like: > exten => 55,1,Goto(,yyy,1) > > but if 55 does not come across and it appears to be no digits > coming across how do I match that that and just start. > > I thought about _X but that says digits. I dont think I am getting any > digits > I just want *anything* coming across to start the call. > > Basically ANY call coming across the trunk just do the same as 55 above. > > Thanks, > > Jerry > > -- > _ > -- 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 -- _ -- 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
Re: [asterisk-users] SIP trunk
Hi Jerry, In article , Jerry Geis wrote: > > It seems I am not getting any digits coming over a SIP trunk. > > How can I match "anything" or "nothing" and start my extension. > > Usually I have something like: > exten => 55,1,Goto(,yyy,1) > > but if 55 does not come across and it appears to be no digits > coming across how do I match that that and just start. > > I thought about _X but that says digits. I dont think I am getting any > digits > I just want *anything* coming across to start the call. > > Basically ANY call coming across the trunk just do the same as 55 above. It sounds like you are thinking of your SIP trunk as if it is a telephone line, when you first pick up the line, and then send digits over it. SIP doesn't work like that. A SIP call is started by an INVITE message that says who the caller is, and what address they want to call. Normally, the address would be something like sip:12...@some.realm.com, where the 12345 would be the digits of the number that would be matched against your dialplan. It doesn't have to be, though, and you could have a call to sip:je...@some.realm.com, which would match the following: exten => jerry,1,Goto(and so on) It is possible, but often not helpful, to have a call that doesn't specify a number, with an address such as sip:some.realm.com, and that should match the "s" extension: exten => s,1,NoOp(Didn't get a number) Maybe that's what is happening in your case, so try adding an "s" extension. Hope this helps, Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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
Re: [asterisk-users] SIP trunk
On Tuesday 26 Jul 2016, Jerry Geis wrote: > It seems I am not getting any digits coming over a SIP trunk. > > How can I match "anything" or "nothing" and start my extension. > > Usually I have something like: > exten => 55,1,Goto(,yyy,1) > > but if 55 does not come across and it appears to be no digits > coming across how do I match that that and just start. Try using extension "s" (for Start). NB: Take care not to include another "s" extension into your context! You can do something like exten => s,1,NoOp(${EXTEN}) which will display in the console, whatever Asterisk thinks was dialled on the far end. If a SIP trunk really is not sending any digits, it can only be used for a single incoming line; the calling party will have to send DTMF to select an extension once your Asterisk answers. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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
[asterisk-users] Yealink T21P E2 bug solved
https://www.mundoopensource.com.br/yealink-t21p_e2-com-bugs-no-firmware-bugs-in-the-yealink-t21p_e2-firmware/ []s Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres -- _ -- 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