Dear Janardhan,
first I would like to thank you for assisting and helping with the SCTP
implementation.
On 08/03/07 00:20, Janardhan Iyengar wrote:
Dear Daniel,
Thank you very much for integrating the code in. This is very good
news indeed!
We will continue to test this code, and add more meaningful use of
SCTP's features as we do more work with it. Who should we consider our
point of contact? Is there one? Or should we just get in touch with
the list again?
the devel(a)openser.org mailing list is the place to discuss
development
issues. It is a list with fast feedback and nice results out of debates
there. Patches should be registered on the tracker.
Your broad experience with SCTP will help a lot. I looked some time ago
to SCTP but the lack of phones supporting the protocol discouraged me
(of course, there is the chicken-egg situation: phone developers need a
server, server developers need a client -- hope this is a good start
point for more SCTP in VoIP). I like the SCTP because the resources used
on server are similar as for UDP, on the other hand, it solves important
issues as transmission reliability and fragmentation.
Another issue I see, an I do not know if applies, does NATs cope with
SCTP? I haven't seen much SOHO routers advertising SCTP.
Another interesting topic would be TLS-SCTP, there is RFC, not time to
study it yet, do you have any experience with it? As I got it from
sip-implementors list, so far the SCTP was used mainly for trunking.
Cheers,
Daniel
Thanks again!
- jana