similar to: Webrtc and adventures with Asterisk 11

Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

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

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

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

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

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

For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms

For the client: JSSIP and Sipml5. If you are going to be coding something up yourself I like the JSSIP 0.5.x javascript interfaces. If you are simply going to use a pre-canned one then sipml5 works pretty well and remembers your settings in localstorage. I haven't used any closed source versions since the above works really well for us. For the server: If you are using Asterisk 1.8

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

users are registering over ws:// but while dialing A -> B , using either jssip/sipml5 I receive an error on B side saying , ice related information is missing , and in INVITE sdp mentioned fields are really missing. Exact error in console is SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd lots of users are using webrtc with jssip/sipml5 + asterisk successfully.

Begin forwarded message: > From: "Tom O'Reilly" <TOreilly at mpegla.com> > Date: May 22, 2010 6:31:50 PM CDT > To: "Dave Johnson" <davefilms.us at gmail.com> > Cc: "Info-web" <Info-web at mpegla.com> > Subject: RE: VP8 > > Dear Mr. Johnson, > > Thank you for writing. We appreciate hearing from you and the >

Patenting a mathematical formula is NOT creating a machine nor is it unique. For example. 2+2=4... apples + apples^2= given outcome. I want to patent this. It's stupid to patent something like that. The same is true for formula algorithms. Algorithms occur in nature. Thus should not be patented. Now, Volley G Mathison inventor of the Electropsychometer had a machine that he could patent. A

"Google will soon make its VP8 video codec open source, we?ve learned from multiple sources. The company is scheduled to officially announce the release at its Google I/O developers conference next month, a source with knowledge of the announcement said. And with that release, Mozilla ? maker of the Firefox browser ? and Google Chrome are expected to also announce support for HTML5 video

Hi all, Sorry for the VP8 question on the Theora list, but I think all the relevant people are here :) I'm using Icecast to distribute some Ogg Theora streams at the moment with a view to adding VP8 along side these in future. Ideally I'd just use Ogg VP8+Vorbis, I know there is a mapping for this already but how much support from the browsers can I expect for this configuration?

2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo (

Thomas, Thank you for your interest in this, you description is as accurate as I can see. > From my perspective your challenges will be to get the containers right. > WebM for audio+video > Ogg for audio > > Also (I'm not that familiar with webRTC) you might need to reencode > to Opus and VP8 in some cases? here is the great news

2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent

Dear Jean-Marc: Would you please send Thierry an ontology mapping table specification for speex according to the format he links to at the end of his message below? Thank you. On Mon, Oct 11, 2010 at 9:54 AM, Thierry MICHEL <tmichel at w3.org> wrote: > Dear James, > > In your latest email to the MAWG response to your comment, you seem to > disagree with the MAWG proposal. >