Hi, I am also trying to integrate sipml5 demo.For that i made some configuration. Call works fine using chrome browser but facing "One way audio issue". And firefox browser not able to originate call.
Here is the my configuration: http://pastebin.com/EtVzK2T2 let me know if i miss something. On Wed, May 21, 2014 at 9:25 PM, Amit Patkar <[email protected]> wrote: > Please check rtp.conf > > Look for stunaddr setting. You can try with google STUN server > stunaddr = stun.l.google.com:19302 > > *Thanks & Regards,* > Amit Patkar > On 5/21/2014 9:13 PM, Gary Shergill wrote: > > Hi again, > > Just noticed this is being sent to the wrong thread... first time using a > mailing list and I just replied to the mail sent by the mailing list for > Amit's reply. Hope this time it works... > > Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side > (I tested using the SIPml demo site and it worked, then realised I was > missing a setting). > > However, the issue still remains where 1000 can not always hear 6901. As > mentioned before, this works only SOMETIMES, and when it does work > asteriskgary.local sees RTP packets coming FROM 192.168.3.131 > (asteriskrtc.local). > > Unsure what would be causing this, because it does work sometimes and doesn't > at others, with no obvious reason either way. > > Thanks again. > > Kind Regards, > > Gary Shergill > > > ----- Original Message ----- > From: "Gary Shergill" <[email protected]> <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> <[email protected]> > Sent: Wednesday, May 21, 2014 3:36:54 PM > Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external > asterisk) > > Hi Amit, > > ICE/STUN is configured correctly. The extension for the webrtc user is > defined in sip.conf on the asteriskrtc.local server. The other user is > defined in Elastix. > > I have "directmedia=no" set for the user on asteriskrtc.local. > > My exact setup/scenario is below: > - asteriskgary.local has a route to dial extensions on my Elastix server. > - asteriskgary.local has a route to dial extensions on asteriskrtc.local > server. > - The call is being originated from asteriskgary.local. The first party is an > extension on asteriskgary.local, the destination party is an extension on my > Elastix server. > > What's happening is as follows (this is a reverse of the previous case as > 6901 is now dialling 1000): > - User on asteriskgary.local places a call to 1000, his number is 6901 > - 6901 answers on the web browser and begins to dial 1000 > - 1000 answers and the call is established correctly > - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...) > - 6901 can NEVER hear 1000 > > key: > 192.168.3.127 - asteriskgary.local > 192.168.3.131 - asteriskrtc.local > 192.168.3.150 - machine running chrome browser where 6901 is logged on > 192.168.3.100 - phone where 1000 is logged on > > (1000 can hear 6901) RTP TRACE ON asteriskrtc.local > .... > Got RTP packet from 192.168.3.150:55148 (type 00, seq 014308, ts > 2304496631, len 000160) > Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054709, ts > 2304496624, len 000160) > > 0x7fe73c021740 -- Probation passed - setting RTP source address to > 192.168.3.127:15942 > Got RTP packet from 192.168.3.127:15942 (type 00, seq 003763, ts 000160, > len 000160) > Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047008, > ts 000160, len 4294967284) > Got RTP packet from 192.168.3.150:55148 (type 00, seq 014309, ts > 2304496791, len 000160) > Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054710, ts > 2304496784, len 000160) > Got RTP packet from 192.168.3.150:55148 (type 00, seq 014310, ts > 2304496951, len 000160) > Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054711, ts > 2304496944, len 000160) > Got RTP packet from 192.168.3.127:15942 (type 00, seq 003764, ts 000320, > len 000160) > Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047009, > ts 000320, len 4294967284) > Got RTP packet from 192.168.3.150:55148 (type 00, seq 014311, ts > 2304497111, len 000160) > Sent RTP packet to 192.168.3.127:15942 (type 00, seq 054712, ts > 2304497104, len 000160) > Got RTP packet from 192.168.3.127:15942 (type 00, seq 003765, ts 000480, > len 000160) > Sent RTP packet to 192.168.3.150:55148 (via ICE) (type 00, seq 047010, > ts 000480, len 4294967284) > .... > > (1000 can hear 6901) RTP TRACE ON asteriskgary.local > ... > Got RTP packet from 192.168.3.131:17836 (type 00, seq 055375, ts > 2304603184, len 000160) > Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004428, ts 106560, > len 000160) > Got RTP packet from 192.168.3.131:17836 (type 00, seq 055376, ts > 2304603344, len 000160) > Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004429, ts 106720, > len 000160) > Got RTP packet from 192.168.3.131:17836 (type 00, seq 055377, ts > 2304603504, len 000160) > Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004430, ts 106880, > len 000160) > Got RTP packet from 192.168.3.131:17836 (type 00, seq 055378, ts > 2304603664, len 000160) > Sent RTP packet to 192.168.3.131:17836 (type 00, seq 004431, ts 107040, > len 000160) > ... > > (no audio) RTP TRACE ON asteriskrtc.local > .... > Got RTP packet from 192.168.3.127:17796 (type 00, seq 035016, ts 000640, > len 000160) > Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060981, > ts 000640, len 4294967284) > Got RTP packet from 192.168.3.127:17796 (type 00, seq 035017, ts 000800, > len 000160) > Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060982, > ts 000800, len 4294967284) > Got RTP packet from 192.168.3.127:17796 (type 00, seq 035018, ts 000960, > len 000160) > Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060983, > ts 000960, len 4294967284) > Got RTP packet from 192.168.3.127:17796 (type 00, seq 035019, ts 001120, > len 000160) > Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060984, > ts 001120, len 4294967284) > Got RTP packet from 192.168.3.127:17796 (type 00, seq 035020, ts 001280, > len 000160) > Sent RTP packet to 192.168.3.150:53684 (via ICE) (type 00, seq 060985, > ts 001280, len 4294967284) > .... > > (no audio) RTP TRACE ON asteriskgary.local > ... > Sent RTP packet to 192.168.3.131:16116 (type 00, seq 035355, ts 054880, > len 000160) > Sent RTP packet to 192.168.3.131:16116 (type 00, seq 035356, ts 055040, > len 000160) > Sent RTP packet to 192.168.3.131:16116 (type 00, seq 035357, ts 055200, > len 000160) > Sent RTP packet to 192.168.3.131:16116 (type 00, seq 035358, ts 055360, > len 000160) > ... > > SIP.conf > [6901] > type=peer > username=6901 > host=dynamic > secret=6901 > qualify=yes > context=webrtc > hasiax = no > hassip = yes > encryption = yes > avpf = yes > icesupport = yes > videosupport=no > directmedia=no > canreinvite=no > > You can see from the trace packets that sometimes asteriskgary.local sees no > packets from asteriskrtc.local, and at the same time the packets on > asteriskrtc.local show half the number of records (there is no "Probation > passed - setting RTP source address to 192.168.3.127:15942 which causes twice > the number of packets, no idea if this is relevant though). > > Please ask if you need anything else. I'm totally stumped with this issue... > Note that on asteriskgary.local ICE is not configured, I wouldn't have though > it would need it as it isn't talking with the webrtc client itself, it is > just talking to an Asterisk server (and that asterisk server is the one which > talks to the webrtc client). > > Thank you. > > Kind Regards, > > Gary Shergill > > > ----- Original Message ----- > From: "Amit Patkar" <[email protected]> <[email protected]> > To: [email protected] > Sent: Wednesday, May 21, 2014 04:41:50 AM > Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external > asterisk) > > Hi Gary > > You need to check if ICE / STUN is configured. > How are these extensions configured? If you are in private network, you > might have to disable DirectMedia / reInvite for calls going between 2 > asterisk boxes. > I hope this helps to resolve your issue. > > *Thanks & Regards,* > Amit Patkar > > > On 5/21/2014 2:26 PM, Gary Shergill wrote: > > Hi, > > I've run into a slight issue when using WebRTC and two Asterisk boxes. > > I am using SIPml as the test WebRTC client. > > My two asterisk boxes, one of them is configured for WebRTC with websockets, > etc (asteriskrtc.local) and the other is just a standard asterisk server > (asteriskgary.local). > > Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to > log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC > user, and vice versa, and all the media flows. > > When I try making a call from the other asterisk server (asteriskgary.local) > to asteriskrtc.local (all routes are set up) I am seeing the following > behaviour: > > - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901 > - 6901 sees the call and has the option to answer > - 6901 answers the call > - 6901 can hear 1000 talking > - 1000 can not hear 6901 > > The weird thing is, sometimes it works, sometimes it doesn't... > > I think it has something to do with the port destination changing when the > call is answered but I'm not sure (wireshark suggests that, as it says "Port > Unreachable"). > > Has anyone tried this before and seen this issue? Or knows why it is and how > to debug it? I can provide any logs required, I have some logs from when it > works and doesn't. > > Thank you for your help. > > Kind Regards, > > Gary Shergill > > > > > > -- > _____________________________________________________________________ > -- 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 > -- Thanks, Bhavik Patel
-- _____________________________________________________________________ -- 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
