[Spce-user] WebRTC not work anymore?

Brian Quandt brian.quandt at gmail.com
Tue Aug 25 18:49:47 EDT 2015


Further testing seems to indicate something else is wrong.

The only way things work is as described below, ie
domain -- per Daniels notes, proxy plain SDP, transport plain rtp,
subscribers -- proxy=never, transport=transparent

If I try to follow Daniels directions, ie
subscribers  proxy additional/only candidate, transport RTP/SAVFP

It does not work (firefox).

does this info help anyone figure out what's up, ie why this works now and
Daniels notes no longer work?
subsriber proxy=never, transport = transport, with domain proxy=plain sdp,
transport plain rtp




On Tue, Aug 25, 2015 at 1:59 PM, Brian Quandt <brian.quandt at gmail.com>
wrote:

> Okay, got it to work at least in one test case.  "I think" the missing
> part was the domain set up part, as outlined in Daniel's notes.   I've read
> his notes on this before... but guessing user-error on not having getting
> everything right between subscriber, domain, and the floating versions of
> firefox/chrome...
>
> Anyways, here's my setup for reference to the future folks.  The ubuntu
> box did not have a camera on it, but the yosemite box did, and was very
> happy to send the video to the ubuntu callee.
>
> You'll note my proxy/protocol settings are different then Daniels.
> Actually, this test that worked was an accident (serendipity), ie my
> subscriber settings got changed in prior tests, and I forgot I changed
> them.  Went to correct the domain settings, didn't look at my old
> subscriber, tested and it worked.  I suspect if I go back to Daniel's notes
> about subscriber settings those too will work, I think the real mistake was
> the domain portion of the setup.
>
>
> Details.
>
> domain: setup per daniels notes
>
> subscriber details:
> user1
> use_rtpproxy -- never
> transport_protocol -- transparent
> Yosemite 10.10.2
> firefox: 39.0
> notes:  tried to use wss for my wsuri, but didn't registered, so fall back
> to plain ws
>
> user2
> use_rtpproxy -- never
> transport_protocol -- transparant
> using trying.jssip.net
> ws uri:  wss://mysipserver:1443/wss/sip/
> ubuntu 14.10
> firefox 35.0.1
> notes:  registered just fine
>
> On Tue, Aug 25, 2015 at 12:23 AM, Daniel Grotti <dgrotti at sipwise.com>
> wrote:
>
>> Hi Brian,
>> no didn't have the chance to do it.
>>
>> --
>> Daniel Grotti
>> VoIP Engineer
>>
>>
>> Sipwise GmbH
>> Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
>>
>> On 08/24/2015 07:37 PM, Brian Quandt wrote:
>> > Daniel,
>> >
>> > Thanks, yes I've seen your post on this before, it was one of the first
>> > things I tried, which lead me to reach out to Jon Bonila and others.
>> >
>> > Have you tried this recently?
>> >
>> > Yours truly,
>> > Brian
>> >
>> >
>> > On Mon, Aug 24, 2015 at 10:28 AM, Daniel Grotti <dgrotti at sipwise.com
>> > <mailto:dgrotti at sipwise.com>> wrote:
>> >
>> >     Hi, I was able to use it here :
>> >
>> >
>> https://www.linkedin.com/pulse/how-enable-webrtc-sipprovider-daniel-grotti?redirectFromSplash=true
>> >
>> >     Daniel
>>
>
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