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 <[email protected]> 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, [email protected] 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> >>>> 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: [email protected] >>>>> Email: [email protected] >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> [email protected] >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> _______________________________________________ >>>> Devel mailing list >>>> [email protected] >>>> 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: [email protected] >>>> Skype: SippySoft >>> >>> _______________________________________________ >>> Devel mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel >> >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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