[OpenSIPS-Users] [OpenSIPS-Devel] Fwd: RTPproxy project

ag at ag-projects.com ag at ag-projects.com
Tue Jun 17 19:11:51 CEST 2014


I think #webrtc is all the rage for all the good or wrong reasons :-)

Is indeed the wrong expectation that a sip server would need to handle this natively but people ask about this and other solutions are there to fill up the gap.

Adrian

On 17 Jun 2014, at 13:17, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:

> Adrian,
> 
> We tried all the time to guide the opensips development (as project) based on the community needs - basically you add features on demand/usage - you mentioned you felt like "left behind feature-wise" - could you mention the features you are missing (especially that you are a foundation member, and we should provide guidance for the project). I'm all ears :).
> 
> It is more or less what I'm doing (as user) with the rtpproxy project - I have the need for some missing features and I'm asking about the future plan.
> 
> Of course, there must be an understanding that different people doing different things may have different needs - this is the beauty of an Open Source project - different people, different needs, all combined into a unitary effort.
> 
> Regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 13.06.2014 20:55, ag at ag-projects.com wrote:
>> Guys,
>> 
>> All these softwares are mature with many years in service both for the media relays and the SIP part. They deal find with most of the expected failures, which is what the customers expect. For the un-expected failures, well the sky if the limit for optimising with infinite cost/benefit ratio. I personally did not hear my customers asking for any more resilience or scalability for the media relay component, so I stopped optimising long time ago.
>> 
>> A better question is where would OpenSIPS project go next, beyond optimisations, as the outside world does not stay still and the perception of some of my customers is that we are being left behind feature-wise.
>> 
>> Adrian
>> 
>> On 13 Jun 2014, at 14:18, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
>> 
>>> Hi Maxim,
>>> 
>>> It is good to know about the rtp_cluster, but aside simplifying things, it does not bring any new functionality - the LB and failover between RTPproxy nodes can be done now in OpenSIPS module .
>>> The most challenging thing we are looking at is the ability to move calls between different instances of RTPP (for HA purposes)..or some restart persistence for the sessions - without something like that it's very hard to deal with SW/HW failures ; there are ways to go around for scheduled stops/restarts (maintenance), but non for unexpected failures.    
>>> 
>>> Thanks and Regards,
>>>  Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>> On 13.06.2014 00:36, Maxim Sobolev wrote:
>>>> Brett, on the HA/carrier-grade side there is little-advertized middle-layer component called "rtp_cluster", which in essence is load-balancing, transparent dispatcher that can be inserted in between some call-controlling component like OpenSIPS or Sippy B2BUA and bunch of RTPP instances running on the same or multiple nodes. From the point of view of that OpenSIPS it's just another RTPP instance.
>>>> 
>>>> And it handles all logic necessary to load-balance incoming requests between online instances plus it can handle dynamic re-confiduration of the cluster and track individual nodes going up and down. The code is pretty usable, we have it deployed for several customers and it's being actively developed as well. We have it working reliably controlling up to 30-40 RTPP instances scattered over at least 5 nodes.
>>>> 
>>>> http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/
>>>> 
>>>> We have at least one pretty well known service provider whose name starts with capital V using it in combination with OpenSIPS to load balance RTP traffic                   via bunch of Amazon EC2 instances.
>>>> 
>>>> 
>>>> On Tue, May 27, 2014 at 6:52 AM, Brett Nemeroff <brett at nemeroff.com> wrote:
>>>> Just wanted to add my 0.02 here.. 
>>>> 
>>>> I totally agree with Bogdan. For the applications where opensips + a RTP relay make sense, HA and persistence are much more important. 
>>>> 
>>>> WebRTC and ICE are kinda applications in of themselves. And although these applications are going to grow in popularity, the "legacy" needs for an RTP relay are still massively prevalent in the space. A general push towards "Carrier Grade", resiliency and redundancy I think is much better for the project as a whole. 
>>>> 
>>>> Not only that, consider that applications requiring ICE or WebRTC will greatly benefit from HA / persistence, but not so much the other way around :) 
>>>> 
>>>> YMMV
>>>> 
>>>> -Brett
>>>> 
>>>> 
>>>> 
>>>> On Sun, May 25, 2014 at 6:30 AM, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
>>>> Hello,
>>>> 
>>>> As always, the truth is in the middle.
>>>> 
>>>> I agree RTPP is behind on certain things (and this is why we want to do them), but on the other hand it is a good platform with other good features (missing on the other relays). RTPP has better ability in individually controlling the stream (audio /video), ability to set timeouts and onhold with no conflicts, ability to generates events on timeout, more flexibility in handling symmetric / asymmetric NATs, ability to do media injection (playback), ability to do call recording
>>>> 
>>>> What neither  mediaproxy, nor rtpengine have is a mechanism for implementing RTP failover (for ongoing calls) or restart persistence . This is something we want to look into. I would love to have ICE and WebRTC on my media relay, for the HA and persistence are more important I would say.
>>>> 
>>>> Regards,
>>>>  Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developer
>>>> http://www.opensips-solutions.com
>>>> On 24.05.2014 01:59, Muhammad Shahzad Shafi wrote:
>>>>> To be honest, i have stopped using rtpproxy for over 2 years now. It is not evolving as fast as it should be, specially in the context of ICE and WebRTC technologies.
>>>>> 
>>>>> I would like to suggest that opensips team should consider adding support for rtpengine from SIPWise,
>>>>> 
>>>>> https://github.com/sipwise/rtpengine
>>>>> 
>>>>> For now mediaproxy from AG Projects is the only good choice for handling media in opensips with ICE support (though it still lacks WebRTC features).
>>>>> 
>>>>> Thank you.
>>>>> 
>>>>>  
>>>>> On 2014-05-23 14:55, Bogdan-Andrei Iancu wrote:
>>>>> 
>>>>>> Going for a public exposure on this question to Maxim, maybe we will get an answer here.
>>>>>> 
>>>>>> 
>>>>>> -------- Original Message --------
>>>>>> Subject:	RTPproxy project
>>>>>> Date:	Mon, 14 Apr 2014 15:03:31 +0300
>>>>>> From:	Bogdan-Andrei Iancu
>>>>>> To:	Maxim Sobolev
>>>>>> CC:	Razvan Crainea
>>>>>> 
>>>>>> 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
>>>>>> 
>>>>>> 
>>>>> -- 
>>>>> Mit freundlichen Grüßen
>>>>> Muhammad Shahzad
>>>>> -----------------------------------
>>>>> CISCO Rich Media Communication Specialist (CRMCS)
>>>>> CISCO Certified Network Associate (CCNA)
>>>>> Cell: +49 176 99 83 10 85
>>>>> MSN: shari_786pk at hotmail.com
>>>>> Email: shaheryarkh at googlemail.com
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Devel mailing list
>>>> Devel at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>>>> 
>>>> 
>>>> 
>>>> 
>>>> -- 
>>>> 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
>>> 
>>> _______________________________________________
>>> Devel mailing list
>>> Devel at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>> 
>> 
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20140617/5cce9aee/attachment-0001.htm>


More information about the Users mailing list