Hello,
courtesy of Glenn Marmon, Connecticut College Class of 2007, who submitted the patch, OpenSER includes now support for SCTP (Stream Control Transmission Protocol). It completes the list of the possible IP transport layers that could be used with SIP in OpenSER.
You can find details about SCTP at: http://www.sctp.org/
Basically, it is a reliable transport protocol, like TCP, but provides a number of functions that are critical for telephony signaling transport (read more about the benefits: http://www.ietf.org/rfc/rfc3286.txt).
If you have the chance to own a SCTP-aware SIP phone and can test, please provide us feedback on the mailing lists -- also, let us know the name and type of any SIP phone with SCTP support that you know. Hope SCTP will gain more and more space, as it seems to be the right protocol for VoIP/IM&P signaling, where UDP cannot cope very well with huge messages and TCP adds too much overhead.
Cheers, Daniel
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?
Thanks again! - jana
On Thursday 02 August 2007, 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?
Hello Janardhan,
nice to hear that you plan to further extend the SCTP feature set!
Discussion about new patches or extensions of features should take place at the devel list. Actual code should be submitted to the project patch tracker: http://sourceforge.net/tracker/?atid=743022&group_id=139143
This way no code submission will be lost or delayed because of vactation or other responsibilities of developers.
Best regards,
Henning
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@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
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Daniel-Constantin Mierla -
Henning Westerholt -
Janardhan Iyengar