[OpenSIPS-Users] RTPproxy project

Fri Jun 20 22:45:02 CEST 2014

Hey Bogdan, thanks for sharing your ideas for us. In fact all of those
items that you have listed are already on drawing board for the next 2-3
months development here:

  - Send timeout notifications to different OpenSIPS servers (more than one)
  - Different timeout values for early media and established calls (longer
for early, shorter for established)
  - Play music on hold in early media state
  - Detect on-hold and disable timeouts (search different solution here)
  - Do not send media timeout if other sessions are active (video and
audio)
  - In bridge mode asymmetric should not be always assumed
  - Cache played files instead of reading them from the disk all the time

I am particularly interested in the timeout notifications and cache for
playing files, so maybe you can start with forking out main branch in the
github and pushing your patches there? For playing cache, I have been
planning to use mmap() and refcounting, so I am particularly curious which
path did you take. We need this in order so that we can use rtpproxy to
generate test streams for other rtpproxy, or maybe even to itself. I have
started some automated regression testing suite here we will be pushing it
our pretty soon.

On Fri, Jun 20, 2014 at 2:22 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

>  Hello Maxim, Hello Jev,
>
> Sorry for taking so long to answer to these emails.
>
> I'm really glad to find out that the rtpproxy project is actually moving
> along and even more, evolving - it is a critical component in our platforms
> (and for most OpenSIPS deployments) and we got a bit concerned about what
> is going on with rtpp. To be honest, we had on the table the possibility to
> fork it and continue by ourselves - but I do not want to re-invent the
> wheel or to pollute the environment with yet another relay relaying tool
> (anyhow, there is this rtpengine stuff popping around lately )
>
> We will be more than happy to get involved - as ideas, experience and work
> - in the rtpproxy evolution ; of course, if you guys are willing to accept
> it :).  One again , rtpproxy is too important to us to stay neutral and
> lately there are more and more features touching both SIP and RTP ....so
> there is a strong need for a better integration between OpenSIPS and
> RTPProxy, IMHO.
>
> Now, technically speaking, the kind of problems we mainly faced are (a)
> scaling with HW (especially with the old single threaded model), (b)
> redundancy and (c) controlling streams (multiple streams audio/video in the
> same SIP session, on-hold, etc).
>
> What we did (and have as patches):
>   - Send timeout notifications to different OpenSIPS servers (more than
> one)
>   - Different timeout values for early media and established calls (longer
> for early, shorter for established)
>   - Play music on hold in early media state
>   - Detect on-hold and disable timeouts (search different solution here)
>   - Do not send media timeout if other sessions are active (video and
> audio)
>   - In bridge mode asymmetric should not be always assumed
>   - Cache played files instead of reading them from the disk all the time
>
>
> Also we are looking into new features (things that we can work together) :
>   - better structuring between sessions and streams
>   - Send timeout notifications over UDP
>   - Force specific ports in reply, if possible
>   - Failover support
>   - Provide statistics per session (even ended) back to OpenSIPS
>   - Restart persistent
>   - Change learning period (possibly linked with on-hold media disable)
>   - ICE support
>   - SRTP to RTP conversion
>
> Definitly we can look into transcoding part too - what we did is for
> Sangoma cards (so HW transcoding, not SW).
>
>
>
> So, we will look into the new work you guys did on rtpproxy - to have a
> starting point for the future planning. After that, if you agree on having
> us contributing to the rtpproxy, we can get involved in planning and actual
> development.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 20.06.2014 02:16, Jev Björsell wrote:
>
> Hi Guys,
>
>  Some updates on the rtpproxy project;
>
>  We have now moved the rtpproxy project from sourceforge to github
> http://github.com/sippy/rtpproxy
>
>  This change should make the project more visibility & and transparency.
> Please feel free to create Issues for feature requests and bugs, and of
> course Pull Requests are appreciated! :)
>
>  We have also moved the mailing list over to Google Groups:
> https://groups.google.com/forum/#!forum/rtpproxy
>
>  We will do a maintenance release - version 1.3, and  Max is busy working
> on a 2.0 release, which has some significant improvements to jitter
> characteristics,  and performance.
>
>  Best Regards,
> -Jev
>
>
>
> On Mon, Jun 9, 2014 at 8:25 AM, Maxim Sobolev <sobomax at sippysoft.com>
> wrote:
>
>>   Hey Bogdan, sorry for missing your message. The mail traffic these
>> days is insane, so it's hard to keep atop of all issues.
>>
>>  We are working behind the scene on what would become rtpproxy 2.0, the
>> code is pretty stable and we have it deployed in like 30-40 places. The
>> main changes are in the timing loop, which improves the jitter
>> significantly and recently we've also split UDP sending code into its own
>> thread(s). That code is available here:
>> https://bitbucket.org/sippysoft/rtpproxy. It's only tested to compile on
>> FreeBSD, but it should not be difficult to compile it on Linux. This
>> basically pushes it to the limits of what's possible to achieve with the
>> standard POSIX facilities. We've been able to push 16-core machine up to
>> 400KPPS in and 400KPPS out with it, all the way up to 90% CPU, while the
>> older version started choking at about 30%. Our plan is to tie few loose
>> ends and push it out to the official repo as a basis for 2.0.
>>
>>  Beyond 2.0, there is another project in progress that is using novel
>> netmap framework to overcome performance issues of the traditional
>> kernel-based socket API. This potentially would allow us to increase
>> capacity at least 5 times on the comparable hardware. The framework itself
>> is pretty low-level, so I am working on a library that would allow it to be
>> more easily integrated into an app. The WiP code is here.
>> https://bitbucket.org/sobomax/libsinet.
>>
>>  Another direction that we are going to explore is to add transcoding
>> support. We have 2 cards in our lab now and setting up the devtesting
>> system just today. I've heard that you have done some work in this
>> direction, so if you want to share something with us, we would be very
>> interested to look at those patches.
>>
>>  On the open-source side we plan to move the project into some modern
>> project management facility, the favorite being github. My colleague Jev is
>> driving this change.
>>
>>  In general I don't mind giving you or anyone else from the OpenSIPS team
>> read-write access to repository if you feel like integrating some of your
>> patches.
>>
>>
>> On Mon, Apr 14, 2014 at 5:03 AM, Bogdan-Andrei Iancu <bogdan at opensips.org
>> > wrote:
>>
>>> Hello Maxim,
>>>
>>> Long time, no talks, but I hope everything is fine on your side.
>>>
>>> I'm reaching you in order to ask about your future plans in regards to
>>> the rtpproxy project? We see no much activity around it and other media
>>> relays are popping around.
>>>
>>> RTPP is an essential component for us, we invested a lot of work, we
>>> have many patches (extensions) for it (which we want to push to the public
>>> tree, but there is no answer on this) and we are also looking for investing
>>> a lot into big future plans (as adding more functionalities).
>>>
>>> Now, my question is - what is your commitment and disponibility for the
>>> RTPP project ? depending on that we what to re-position ourselves, as we do
>>> not want to waste time and work on things which are out of control.
>>>
>>> Best regards,
>>>
>>> --
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>>
>>>
>>
>>
>> --
>> Maksym Sobolyev
>> Sippy Software, Inc.
>> Internet Telephony (VoIP) Experts
>> Tel (Canada): +1-778-783-0474
>> Tel (Toll-Free): +1-855-747-7779
>> Fax: +1-866-857-6942
>> Web: http://www.sippysoft.com
>> MSN: sales at sippysoft.com
>> Skype: SippySoft
>>
>
>
>

-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
Tel (Canada): +1-778-783-0474
Tel (Toll-Free): +1-855-747-7779
Fax: +1-866-857-6942
Web: http://www.sippysoft.com
MSN: sales at sippysoft.com
Skype: SippySoft
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