On 07 Jan 2014, at 15:55, SVN commits to the Digium repositories
<[email protected]> wrote:
> if (endpoint->nat.rewrite_contact && (contact =
> pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL)) &&
> - (PJSIP_URI_SCHEME_IS_SIP(contact->uri) ||
> PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
> + !contact->star && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) ||
> PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
> pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
>
> pj_cstr(&uri->host, rdata->pkt_info.src_name);
>
Seems like this code assumes that SIP only have SIP: and SIPS: as URI's. We
should actually be quite transparent in the schemas supported - especially
proxys but also b2bua's like Asterisk. Tel: uri's are not unknown. In the
security area we are discussing improved end-2-end security which may end up
using a new SIP uri.
Instead of testing with two functions for two classes of schemas a table could
be used? That would be more extensible. And please implement tel: support :-)
/O
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