On 07/12/2012 03:53 PM, Benny Amorsen wrote:
chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial.It would be greatly appreciated :) It is low priority for the Asterisk project, as there are always workarounds.
I've just looked into this a bit, and I don't see how using connect() would actually solve the problem. If we receive a UDP datagram from a SIP endpoint, we could use socket() and connect() to create a socket specifically for sending to (and receiving from) that endpoint in the future, but we can't specify the source address to be used by that socket. The only way I know of to specify the source address for outbound packets is to use a raw socket and compose the IP header ourselves, which would be overkill.
Benny, are you aware of some other method to accomplish this? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming@xxxxxxxxxx | SIP: kpfleming@xxxxxxxxxx | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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