[asterisk-dev] [asterisk-commits] russell: branch bbryant/sip-tcptls r73108 - /team/bbryant/sip-tcptls/channels/

Jeffrey C. Ollie jeff at ocjtech.us
Tue Jul 3 11:17:19 CDT 2007


On Tue, 2007-07-03 at 16:00 +0000, SVN commits to the Asterisk project
wrote:
>  
> +/*!< Define some SIP transports */
> +enum sip_transport {
> +	SIP_TRANSPORT_UDP = 1,
> +	SIP_TRANSPORT_TCP = 1 << 1,
> +	SIP_TRANSPORT_TLS = 1 << 2
> +};

Note that there is some work afoot to provide more options for
transporting SIP:

http://tools.ietf.org/html/draft-jennings-sip-dtls-04

Although I don't expect support to be coded now, perhaps it would be
good to have a more flexible approach to specifying the transport.

Also, is there a point to defining it like a bitfield?  Unless you are
going to be 'or'ing and 'and'ing them together I think it makes more
sense to leave them as integers.

Jeff
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20070703/932572f2/attachment.pgp 


More information about the asterisk-dev mailing list