[asterisk-dev] Asterisk TCP

[asterisk_dev]

Great, so, how can I start? I think a good start would be looking at your 
chan_sip3. I've never been in an open-source project, so I'm a little lost 
with that, it would be nice if someone can help me with that.


----- Original Message ----- 
From: "Olle E Johansson" <oej at edvina.net>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Cc: "Olle E Johansson" <oej at edvina.net>
Sent: Sunday, February 05, 2006 12:07 PM
Subject: Re: [asterisk-dev] Asterisk TCP


>
> 5 feb 2006 kl. 11.33 skrev asterisk_dev:
>
>> Hi;
>>
>> I've seen on bugs.digium.com the main developer is going to be  busy, and 
>> I'd like to take over this. I've experience developing  protocols over 
>> Linux (have done OSPF and BGP implementations). Do  everybody agree?
> Great.
>
> The patch in the bug tracker is not really a good start, but can be  seen 
> more as a proof-of-concept experiment.
>
> A good start would be helping analyzing what needs to be done to  support 
> SIP/TCP, how we can handle switching from UDP
> to TCP in midcall, how we can make the channel aware of the socket 
> bindings and stop retransmissions and other things
> that the TCP layer will handle. We also need to handle closing of  sockets 
> for registrations dynamically, so we have a
> mechanism that somehow caches connections and manages them properly,  I 
> would guess the same thing as any
> http server has to do with clients that wants to keep connections open.
>
> The TCP implementation requires quite a lot of work to be done  properly 
> and that patch implements some, but very
> little of what is required. SIP over TCP needs the new socket  interface 
> that Kevin has on hist to-do list and a serious
> change of chan_sip internals, which is on my to-do list under the  project 
> name "chan_sip3".
>
> Both the IPv6 and the SIPS support puts a lot of requirements on the  sip 
> channel code.
>
> When that groundwork is done, it's time for TCP and TLS.
>
> /Olle
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