[OpenSIPS-Devel] New OpenXCAP release 2.2.0

2014-12-03 Thread ag
Changelog

openxcap (2.2.0)

  * Add a 30 second timeout to avoid keeping TCP lingering connections
  * Removed deprecated port setting
  * Raise open file descriptor limit on start
  * Lower HTTP input timeout to 30 seconds
  * Dropped Python  2.7 support

To upgrade or install see http://openxcap.org

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Re: [OpenSIPS-Devel] [opensips] Changes to stop keepalives on unregisters (#366)

2014-11-25 Thread ag
On 25 Nov 2014, at 08:04, David Sanders notificati...@github.com wrote:
 @saghul, fair points.
 
 This problem is known, but it never seemed worth fixing because it would 
 involve keeping track of the registered AoR for NAT_Contact.
 
 It was worth fixing for us, hence this fix which worked for our needs. It 
 would seem that the multiple accounts from the same IP and port would be more 
 of a corner case than the main use case.
 

Well, your corner case may be other's main usage scenario.

My main usage is multiple accounts on the same client and I will not be 
reachable at any of them by using the current patch. We have users on 
sip2sip.info using several accounts registered at the same time from the same 
end-point as a normal use case.

The patch should improve things and fix also your problem but without 
sacrificing existing functionality that you don't care much about, others do.

Adrian
 In ensuring the multiple accounts case works you're sacrificing the main use 
 case. Without this change keepalives just continue for user agents that 
 unregister after registering for a long period.
 
 In a production environment with ~70,000 user agents registering and 
 unregistering constantly this was basically unacceptable behavior due to the 
 continued keepalives.
 

 At the least there should probably be a high-visibility note in the 
 documentation for the module that explains that keepalives will continue to a 
 user agent after unregister.
 
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Re: [OpenSIPS-Devel] Devel Digest, Vol 72, Issue 27

2014-06-25 Thread ag
Sounds like a generic tunnelling technique that has little to do with SIP or 
media. Why can’t you just use OpenVPN or similar ?

Adrian

On 25 Jun 2014, at 08:28, kaushik parmar androidj...@gmail.com wrote:

 Hi,
 
 I am encrypting SIP and RTP message before sending it from sip mobile dialer 
 and it sends UDP packet over network. So no one can know about the type (SIP 
 or RTP) of packet until we decrypt it. This is for secured call and also 
 solution for voip blocked countries.
 
 I want to know which file or module is used in opensips to get and send udp 
 packets? when opensips receives message , i will decrypt it and before send 
 response to mobile dialer , i will encrypt the message. Same for rtpproxy 
 server.
 
 
 
 
 On Wed, Jun 25, 2014 at 2:41 PM, Bogdan-Andrei Iancu bog...@opensips.org 
 wrote:
 Hi Kaushik,
 
 So the while SIP package is encrypted . It is not easy to add hooks before 
 the SIP stack (between transport layer and SIP stack), but can be done - 
 could you provide more details how the encryption / decryption works, if over 
 UDP or TCP, etc ?
 
 Regards,
  Bogdan-Andrei Iancu
 OpenSIPS Founder and Developer
 http://www.opensips-solutions.com
 On 25.06.2014 09:26, kaushik parmar wrote:
 Hi Adrian,
 
 It is not OTR. Actually we have own algorithm for encryption and decryption 
 of sip and rtp packets. We implemented it in our SIP mobile dialer. Now we 
 need to implement it on proxy server. I want to add encryption and 
 decryption code in opensips (and rtpproxy) so opensips (rtpproxy) can come 
 to know that it is SIP and rtp packets. Can you please tell me where should 
 i add this code in opensips?  I am searching for file where opensips getting 
 sip messages and from where it sends/forward sip messages.
 
 
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1. Re: [OpenSIPS-Users] Encrypt and Decrypt sip  signals
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 --
 
 Message: 1
 Date: Mon, 23 Jun 2014 16:11:46 -0300
 From: a...@ag-projects.com
 Subject: Re: [OpenSIPS-Devel] [OpenSIPS-Users] Encrypt and Decrypt sip
 signals
 To: OpenSIPS users mailling list us...@lists.opensips.org
 Cc: OpenSIPS devel mailling list devel@lists.opensips.org
 Message-ID: 2ea9531b-f5b8-42ef-b513-395b6a493...@ag-projects.com
 Content-Type: text/plain; charset=us-ascii
 
 Perhaps is using OTR? In this case the encryption is end-to-end and cannot 
 be handled by an intermediary as it defies the purpose.
 
 Adrian
 
 On 23 Jun 2014, at 04:24, Olle E. Johansson o...@edvina.net wrote:
 
 
  On 23 Jun 2014, at 09:12, kaushik parmar androidj...@gmail.com wrote:
 
  Hello All,
 
  My Android mobile SIP Dialer is sending Encrypted SIP messages
  Is it actually using S/MIME to decrypt on a per-message basis or do you 
  mean it's using TLS as a transport?
 
  /O
 
  and i want to decrypt that SIP message on opensips proxy server. Opensips 
  server will Decrypt the sip request and forward it to my voip server. 
  Same way it will take sip request of voip switch , Encrypt it and send 
  Encrypted SIP request to Android mobile Application.
 
  Can anyone tell me where should i write Encryption and Decryption code in 
  opensips? Is there any particular file in which i can write my encryption 
  code?
 
 
  --
  Kind regards,
 
  Kaushik Parmar
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Re: [OpenSIPS-Devel] [OpenSIPS-Users] Encrypt and Decrypt sip signals

2014-06-23 Thread ag
Perhaps is using OTR? In this case the encryption is end-to-end and cannot be 
handled by an intermediary as it defies the purpose.

Adrian

On 23 Jun 2014, at 04:24, Olle E. Johansson o...@edvina.net wrote:

 
 On 23 Jun 2014, at 09:12, kaushik parmar androidj...@gmail.com wrote:
 
 Hello All,
 
 My Android mobile SIP Dialer is sending Encrypted SIP messages
 Is it actually using S/MIME to decrypt on a per-message basis or do you mean 
 it's using TLS as a transport?
 
 /O
 
 and i want to decrypt that SIP message on opensips proxy server. Opensips 
 server will Decrypt the sip request and forward it to my voip server. Same 
 way it will take sip request of voip switch , Encrypt it and send Encrypted 
 SIP request to Android mobile Application.
 
 Can anyone tell me where should i write Encryption and Decryption code in 
 opensips? Is there any particular file in which i can write my encryption 
 code?
 
 
 -- 
 Kind regards,
 
 Kaushik Parmar
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Re: [OpenSIPS-Devel] [OpenSIPS-Users] Fwd: RTPproxy project

2014-06-17 Thread ag
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 bog...@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, a...@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 bog...@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 br...@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 bog...@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

Re: [OpenSIPS-Devel] [opensips] [RFC] An initial attempt of porting rtpproxy-ng module from your twin project to OpenSIPS. (#152)

2014-06-17 Thread ag
I guess it is reasonable that nobody expects a developer to stretch to manage 
his stuff across external forks. I am not maintaining MediaProxy in Kamailio 
project either, if there are volunteers there it is fine but if they are not, 
then end-users have indeed a problem with any such fork.

I guess Bogdan’s questions is very legitimate, who is going to maintain this 
new module?

Adrian  

On 17 Jun 2014, at 13:05, Richard Fuchs notificati...@github.com wrote:

 Just to clear things up, I had no part in porting this module to opensips 
 other than the original implementation from which it is derived.
 
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Re: [OpenSIPS-Devel] [OpenSIPS-Users] Fwd: RTPproxy project

2014-05-28 Thread ag
Well, the way we implemented ‘persistence’ was by applying a different 
thinking. The goal is to allow live software updates without disrupting 
traffic. With MediaProxy one can shutdown gracefully a relay by allowing it to 
carry on finishing existing calls and then shutdown while the traffic is 
handled by other relays. This way one can upgrade the software on a relay farm 
without dropping a single call.

Adrian

On 28 May 2014, at 03:04, Bogdan-Andrei Iancu bog...@opensips.org wrote:

 Saul,
 
 the carrier grade features are mainly referring to HA and persistenceacross 
 restarts.
 
 Regards,
 
 Bogdan-Andrei Iancu
 OpenSIPS Founder and Developer
 http://www.opensips-solutions.com
 
 On 27.05.2014 23:07, Saúl Ibarra Corretgé wrote:
 On May 27, 2014, at 4:29 PM, Bogdan-Andrei Iancu wrote:
 
 Brett, you put the finger on the wound :)
 
 I looked around to other alternatives (to avoid re-inventing the wheel) - 
 like mediaproxy or rtpengine - and I saw no carrier-grade features in the 
 there  - please correct me if I'm wrong.
 
 I'm looking to see if the problem is correctly identified and if there is a 
 large consent in the community about this need. As we would like to through 
 some resources into this (hopefully other parties too), as ideally we 
 should be going in the right direction :)
 
 What carrier grade features are those?
 
 --
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 AG Projects
 
 
 
 
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Re: [OpenSIPS-Devel] [OpenSIPS-Users] Fwd: RTPproxy project

2014-05-28 Thread ag
Regarding high availability. I admit that in ten years of deployments I have 
never heard a single customer asking for this. They were rather asking for 
having multiple relays in multiple data centers because losing one IP was in 
most cases associated with complete connectivity failure to that data centre 
and a single IP failover was something that even if possible the costs far 
exceeded the benefits. A single IP address going down out of a larger 
connectivity issue context is such a rare occurrence, practically I don’t 
remember ever hearing a customer complaining about such a thing.

In my opinion addressing new features like what rtp engine solves with regards 
to interoperability in more real time scenarios is a smarter investment rather 
than optimising in places were the benefits can be hardly measurable. Not to 
mention that the term carrier-grade” like its predecessor five times 9 
availability is slowly exiting the vocabulary and is being replaced with 
webRTC ready and other newer concepts.

My two cents

Adrian

On 28 May 2014, at 09:35, a...@ag-projects.com wrote:

 Well, the way we implemented ‘persistence’ was by applying a different 
 thinking. The goal is to allow live software updates without disrupting 
 traffic. With MediaProxy one can shutdown gracefully a relay by allowing it 
 to carry on finishing existing calls and then shutdown while the traffic is 
 handled by other relays. This way one can upgrade the software on a relay 
 farm without dropping a single call.
 
 Adrian
 
 On 28 May 2014, at 03:04, Bogdan-Andrei Iancu bog...@opensips.org wrote:
 
 Saul,
 
 the carrier grade features are mainly referring to HA and 
 persistenceacross restarts.
 
 Regards,
 
 Bogdan-Andrei Iancu
 OpenSIPS Founder and Developer
 http://www.opensips-solutions.com
 
 On 27.05.2014 23:07, Saúl Ibarra Corretgé wrote:
 On May 27, 2014, at 4:29 PM, Bogdan-Andrei Iancu wrote:
 
 Brett, you put the finger on the wound :)
 
 I looked around to other alternatives (to avoid re-inventing the wheel) - 
 like mediaproxy or rtpengine - and I saw no carrier-grade features in the 
 there  - please correct me if I'm wrong.
 
 I'm looking to see if the problem is correctly identified and if there is 
 a large consent in the community about this need. As we would like to 
 through some resources into this (hopefully other parties too), as ideally 
 we should be going in the right direction :)
 
 What carrier grade features are those?
 
 --
 Saúl Ibarra Corretgé
 AG Projects
 
 
 
 
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Re: [OpenSIPS-Devel] [OpenSIPS-Users] Fwd: RTPproxy project

2014-05-28 Thread ag
Not me. But most of the people are buying these in flocks!

On 28 May 2014, at 13:26, Bogdan-Andrei Iancu bog...@opensips.org wrote:

 Would you buy a Iphone 6 ready car over a NCAP - proven car ? :)
 
 Thanks and regards,
 
 Bogdan-Andrei Iancu
 OpenSIPS Founder and Developer
 http://www.opensips-solutions.com
 
 On 28.05.2014 17:40, a...@ag-projects.com wrote:
  Not to mention that the term carrier-grade” like its predecessor five 
 times 9 availability is slowly exiting the vocabulary and is being replaced 
 with webRTC ready and other newer concepts.
 
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[OpenSIPS-Devel] New MediaProxy version 2.6.1 released

2014-05-07 Thread ag
Hello,

There is a new MediaProxy release with fixes related to ICE support.

For change log see

http://mediaproxy.ag-projects.com/news/161

Regards,
Adrian

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Re: [OpenSIPS-Devel] [opensips] [RFC] An initial attempt of porting rtpproxy-ng module from your twin project to OpenSIPS. (#152)

2014-03-06 Thread ag
Hi Andreas,

Not sure if you are referring to me but in case you do. I am not demanding any 
apology. Personally, I have no issue with the name you have chosen. If you 
knowingly chose that name mediaproxy-ng which I was using since 2008 and 
expected me to complain about it, is really weird. I am not polling github 
every day just in case something like this happens.

The issue emerged now, in the context of OpenSIPS project where we have an 
obvious overlap of naming and purpose at the same time.

As I mentioned, your software is a great addition and we just asked to 
eliminate any confusion by renaming it.

If you can help with the name change is great, if not then so be it, I don’t 
really care about your choices and I don’t need any apologies from you or 
anybody else.

Adrian

On 06 Mar 2014, at 09:07, Andreas Granig notificati...@github.com wrote:

 Speaking with my @sipwise hat on, there will be a name change of at least the 
 mediaproxy-ng back-end part soon, to clear things up for the future. I 
 understand the concerns of AG in regards to name clashing, and we don't like 
 the confusion either. No plans yet for the rtpproxy-ng module name though, 
 because as @rfuchs pointed out, it's really an evolution of the rtpproxy 
 module, and it can be used as rtpproxy drop-in replacement when switching the 
 back-end.
 
 Since only the module part (rtpproxy-ng) is going to be maintained by you 
 guys (we don't use opensips, so there are no plans to support the module for 
 opensips), you can name it however you like.
 
 However, I find it extremely offensive from this community to have a module 
 pulled into your project, then start accusing the author and demanding an 
 apology for the name, after he's kind enough to chime in and provide an 
 explanation for the naming clashes. The media proxy is available as open 
 source on github since nearly two years, and I would have expected some 
 feedback from AG if there were concerns about the name over this period (it's 
 not that the guys at AG don't know who we are and what we do).
 
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Re: [OpenSIPS-Devel] [opensips] [RFC] An initial attempt of porting rtpproxy-ng module from your twin project to OpenSIPS. (#152)

2014-02-23 Thread ag
Call it fuchs-relay, sipwise-relay or anything else but rtpproxy and mediaproxy 
as they have nothing to do with any of them.

Adrian


Pe 22.02.2014, la 12:13, Richard Fuchs notificati...@github.com a scris:

 Correct, the protocol is different and incompatible, but heavily based on the 
 original rtpproxy protocol in terms of functionality. The protocol offers all 
 the same flags that the original rtpproxy protocol supported, just 
 transported in a different way to allow it to be extended freely. Also, at 
 least in the original rtpproxy-ng module, the function interface is 
 completely identical to the rtpproxy module.
 
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Re: [OpenSIPS-Devel] [opensips] [RFC] An initial attempt of porting rtpproxy-ng module from your twin project to OpenSIPS. (#152)

2014-02-21 Thread ag
Why don’t you simply call it

sipwise-rtp-mediarelay

or something similar so is beyond doubt who made it and where it fits or does 
not fit.

Secondly mediaproxy-ng.org domain name was used by AG Projects for many years 
for hosting our MediaProxy software version.

Regards,
Adrian


On 21 Feb 2014, at 13:59, Richard Fuchs notificati...@github.com wrote:

 As the author of mediaproxy-ng, let me try to clear up the confusion about 
 the naming.
 
 Many moons ago, the team at Sipwise was using a privately developed, closed 
 source RTP proxy. It was designed to be used with the Openser mediaproxy 
 module, and as such we called our project mediaproxy, even though it was 
 completely unrelated to the AG Projects Mediaproxy.
 
 Later on, we decided to redesign our mediaproxy from scratch and eventually 
 make it open source. Thus, mediaproxy-ng was born. At around the same time, 
 we decided to shift our focus away from the Openser mediaproxy module and 
 support the control module rtpproxy instead (even though compatibility with 
 the other module was retained).
 
 Yet again later on, consensus among developers was that the future way to go 
 for media/RTP proxying was to employ a JSON-like control protocol that allows 
 complete rewriting of the entire SDP body. We went ahead and implemented this 
 into mediaproxy-ng. As a new control module was required, we took the old 
 rtpproxy module and modified it. As the new module was (and still is) 
 intended as a drop-in replacement for the rtpproxy module (and not the 
 unrelated mediaproxy module), we called it rtpproxy-ng to make 
 transitioning easier.
 
 Other people have suggested to call the new module mediaproxy-ng instead of 
 rtpproxy-ng, which would be more logical because it was meant to be used 
 with the mediaproxy-ng daemon, but then that would imply that this module 
 somehow is a fork or modification of the old mediaproxy module, which it 
 isn't. All the functions within rtpproxy-ng are taken more or less directly 
 from rtpproxy without even renaming them, so it makes sense to retain 
 rtpproxy in the name of the module.
 
 Also, there's no reason why rtpproxy-ng cannot be used with other RTP/media 
 proxies if they choose to implement this new protocol. By no means is it 
 exclusive to mediaproxy-ng.
 
 So the reasons for the naming are entirely historical. Other than the reasons 
 given for easy transitioning from rtpproxy to rtpproxy-ng, there's no 
 reason why the module can't be renamed to anything else.
 
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Re: [OpenSIPS-Devel] [opensips] [RFC] An initial attempt of porting rtpproxy-ng module from your twin project to OpenSIPS. (#152)

2014-02-21 Thread ag
Hello Richard,

This is not silly. Is very healthy to argue about it, now that the time has 
come. Here it is what you wrote:
Many moons ago, the team at Sipwise was using a privately developed, closed 
source RTP proxy. It was designed to be used with the Openser mediaproxy 
module, and as such we called our project mediaproxy, even though it was 
completely unrelated to the AG Projects Mediaproxy.

Later on, we decided to redesign our mediaproxy from scratch and eventually 
make it open source. Thus, mediaproxy-ng was born. At around the same time, we 
decided to shift our focus away from the Openser mediaproxy module and 
support the control module rtpproxy instead (even though compatibility with 
the other module was retained).

Yet again later on, consensus among developers was that the future way to go 
for media/RTP proxying was to employ a JSON-like control protocol that allows 
complete rewriting of the entire SDP body. We went ahead and implemented this 
into mediaproxy-ng. As a new control module was required, we took the old 
rtpproxy module and modified it. As the new module was (and still is) 
intended as a drop-in replacement for the rtpproxy module (and not the 
unrelated mediaproxy module), we called it rtpproxy-ng to make 
transitioning easier.

Other people have suggested to call the new module mediaproxy-ng instead of 
rtpproxy-ng, which would be more logical because it was meant to be used with 
the mediaproxy-ng daemon, but then that would imply that this module somehow is 
a fork or modification of the old mediaproxy module, which it isn't. All the 
functions within rtpproxy-ng are taken more or less directly from rtpproxy 
without even renaming them, so it makes sense to retain rtpproxy in the name 
of the module.

Also, there's no reason why rtpproxy-ng cannot be used with other RTP/media 
proxies if they choose to implement this new protocol. By no means is it 
exclusive to mediaproxy-ng.“

End of quote.

I am not amused to have to point to it. Building consensus around of chain of 
re-entrant poor choices is not an excuse for not be willing to rename your 
software now. Admit the mistake (it may not even be yours, things just evolved 
to this unfortunate state of affairs) and fix it now.

How about:

My name is Richard Fuchs and I made a new RTP media relay module that makes a 
difference, it does X, Y and Z features than no other modules of OpenSIPS do to 
date. I am aware that the software name was unfortunate as it collides with 
similar implementations but I CAN change it to fix the problem because I am the 
AUTHOR. I can change  the software name to fuchs-relay or whatever name avoids 
collisions. Can it be added to the project, what do you think? 

And the answer would be a big Yes, this is great addition, and welcome. 
Instead, you are trying to justify the chain of bad name choices and claim 
‘consensus’ of developers that are not involved of this project.

Adrian


On 21 Feb 2014, at 22:16, Richard Fuchs notificati...@github.com wrote:

 Actually it's not all that easy. Plus, I find it silly to argue about a mere 
 name like that.
 
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 Reply to this email directly or view it on GitHub.
 
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