Re: [rtcweb] Requesting "SDP or not SDP" debate to be re-opened

[Bossiel thioriguel <[e-mail]>]

Forgot to add:
We're willing to provide an open source gateway to SIP/IMS world and feedbacks for any new implementation to show "how good" or "how bad" it is :)


________________________________
 De : Bossiel thioriguel <[e-mail]>
À : "[e-mail]" <[e-mail]> 
Cc : "[e-mail]" <[e-mail]> 
Envoyé le : Vendredi 21 juin 2013 11h40
Objet : Re: [rtcweb] Requesting "SDP or not SDP" debate to be re-opened
 


Hello,

I'm registered on this group since the beginning but this is my first post on this thread. So, I presente myself: Mamadou DIOP and I'm working for Doubango Telecom where we're building SIP endpoints, gateways, TelePresence/Telemedicine systems... all focused on SIP/IMS/LTE/RCS-e and open source.

What I'm talking about is not just feeling but something I've experienced.

Using the current WebRTC we have managed to *easily* build almost all kind of applications: click-to-call, SIP/IMS clients, gateways to PSTN, MCUs, Telemedicine systems...and haven't seen any major issue. It's true that it's not natural to "hack" a blob SDP to implement features like hold/resume, media update, early media ... but it works and there are demo applications showing it. If there is something more beautiful we just want to see it in action and test it.

Many participants here have said that what they want is something close to CU-RTC-WEB. Don't really know if they tried to build applications using it or not but in my case I have.
My reference: http://html5labs.interoperabilitybridges.com/prototypes/cu-rtc-web-roaming/cu-rtc-web-roaming/info
First on Windows 8 but haven't gone far as there is no documentation to get started. Then, OSX and luckily there was a readme with two links for testing (only one work). You need to open 3 pages (1 master, 2 slaves) and check "send audio" on both slaves to header sound. Many javascript files and no documentation. It's said on these blogs that interop with SIP networks is easy but it's not my feeling ...I just want to see one :)

I don't really understand the issue with the O/A model. SDP or not SDP you'll always offer something and answer something. I'm I missing?

For the current WebRTC, Google open sourced their engine, produced drafts, a working implementation in chrome, a mailing-list to help developers, demo applications, documentation... we just want to see the same from any company asking to rewrite everything.

I'm not saying the current WebRTC implementation is perfect but I have seen my 14 year old nephew developing an audio/video chat for his homework :)

Regards


________________________________
 De : Iñaki Baz Castillo <[e-mail]>
À : Emil Ivov <[e-mail]> 
Cc : "[e-mail]" <[e-mail]> 
Envoyé le : Vendredi 21 juin 2013 1h24
Objet : Re: [rtcweb] Requesting "SDP or not SDP" debate to be re-opened
 

2013/6/21 Emil Ivov <[e-mail]>:
>
> On 20.06.13, 23:49, Iñaki Baz Castillo wrote:
>>
>> In JsSIP we are getting frustrated trying to implement the "hold" /
>> "unhold" feature because it requires SDP parsing and mangling. Sending
>> a re-INVITE with a modified SDP (now with a
 video track enabled) seems
>> to work (after lot of pain) but we still miss a reliable API to know
>> what the new SDP means. Instead we need to parse the SDP to detect
>> global (or per m=) line attributes like "a=inactive" or "a=sendonly"
>> etc etc. It's really painful.
>
>
> I am having a problem following what you are trying to achieve here. In
> JsSIP you seem to be going for a full SIP implementation in the browser. If
> this is true and if this WG decides to remove SDP from the API surface, then
> you would need to completely parse SDP in the JS and then convert it into
> API calls. Similarly, when creating offers and answers you would need to
> construct SDP all by yourself.

And we will do it very happily because then we will know what
*exactly* we are sending on-the-wire.




> So I am not sure why the SDP parsing in the current
 situation is so much of
> a blocker for your use case.

Because regardless I am a SIP-guy, I understand that the main mission
of WebRTC is to provide realtime communications *for* the WWW, and not
to enable a new interface for like-telephony-bussines.

Today I'm doing SIP. Tomorrow I may be doing
[[put_here_a_future_RTC_protocol_not_based_on_SDP]] and then I don't
want to be constrained by decisions made today that force any future
RTC protocol to deal with SDP O/A model.

:)



>> BTW I don't know wheter you support PlanA, PlanB or NoPlan, but I did
>> a question (in this case about NoPlan) for which I got no response,
>> and honestly I would like to see it replied regardless the solution
>> uses PlanA, PlanB or NoPlan model:
>>
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg07871.html
>>
> The other option would be indeed to do the same thing in JS. I believe this
> is JsSIP's use case. In that case however, regardless of whether you choose
> Plan A, Plan B, No Plan or CU-RTC-Web, you will inevitably be exposed to a
> fair amount of complexity, parsing and JS magic.
>
> You are, after all, building a SIP stack.

Yes, but JsSIP creates its own SIP messages to be sent in the wire, so
we have full control over *what* we create and send. Those SIP
messages are not provided by the WebRTC API. But for the SDP
component, JsSIP retreives a SDP blob string from the PC.







>
> In the above mail you also say:
>
>> Another example:
>>
>> * I am a powerful SIP conference server which properly implements
>> WebRTC. I initiate
 a call to 5 users (running JS SIP app in their
>> browsers). The initial INVITE has SSRC/MSID fields in the SDP
>> identifying all the participants, am I right?
>
>
> No, with No Plan there are no SSRCs and MSIDs in the SDP that comes from the
> browser.

OK


>> * Later, during the conference, I call to another 6th participant and
>> enter him into the conference, so I need to send a re-INVITE to every
>> participant with a modified version of the SDP (note that this is SIP
>> protocol, so I need to use SIP messages to carry the new info about
>> SSRC/MSID and so on).
>
>
> That's the thing. You don't need that. In Jitsi we do exactly this operation
> with no Offer/Answer signalling. RTP carries enough information to process
> streams and we use upper layer signalling (4575) for things such as mapping
> SSRCs to users
 and announcing current participant list.

That is much better than Plan A and Plan B.



BTW: What would happen in NoPlan if the remote (i.e. a SIP
gateway/endpoing) sends you a re-INVITE for "hold" purposes and you
pass the SDP to your PC? or you should not pass the SDP to your PC?
and if so, what about if the SDP contains updated ICE candidates due
to remote peer network mobility?



Thanks a lot for your response.

--
Iñaki Baz Castillo
<[e-mail]>
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