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Re: [osip-dev] RE-INVITE not supported?

From: Aymeric Moizard
Subject: Re: [osip-dev] RE-INVITE not supported?
Date: Tue, 29 Nov 2016 23:16:05 +0100


I'm not experienced at all with "overlapping". 

However, according to me, it is very clear that sending a re-INVITE before
receiving the final answer to initial INVITE is forbidden by SIP/rfc3261.

That said, I've been reading https://www.ietf.org/rfc/rfc3578.txt and it
looks to be specified that it is possible to send new INVITE with same
"from-tag" and "call-id":

   The new INVITE has the same
   Call-ID and the same From header field including the tag as the first
   INVITE sent, but has an updated Request-URI.  The new Request-URI
   contains all the digits received so far.  The To header field of the
   new INVITE contains all the digits as well, but has no tag.

This will requires evolutions in libexosip2. Most probably, no change in libosip2... ;)

Do not hesitate to contact me for those evolutions/patchs!


2016-11-29 22:25 GMT+01:00 Roger Schreiter <address@hidden>:

how does libosip2 handle overlap dialling for incoming calls?
Is it necessary, that libosip sends "number incomplete" before the subsequent INVITE arrives (sending more digits)?

I do not know where it is specified (RFC?), but imho the SIP gateway should be able to accept the subsequent INVITE, even if it did not yet send the number incomplete message (e.g. because database lookup is still taking time). Maybe this behaviour is even not specified in RFC, but a common workaround in some gateways, which are used for gatewaying to PSTN in countries, where number length is variable.
(In asterisk this behaviour can be switched on by a conf file variable. Similar in Dialogic's IMG.)

I'm asking, because I'm working on a project, which will need to support overlap dialling, and I hope, I won't run into troubles. The project is based on libosip2, thus I will have to find a solution.


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