Merged revisions 315894 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r315894 | mnicholson | 2011-04-27 14:14:27 -0500 (Wed, 27 Apr 2011) | 28 lines
  
  Merged revisions 315893 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
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    r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
    
    Merged revisions 315891 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
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      r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
      
      Fix our compliance with RFC 3261 section 18.2.2.
      
      This change optimizes the free_via() function and removes some redundant null
      checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
      the port specified in the Via header for routing responses (even when maddr is
      not set). Also the htons() function is now used when setting the port.
      Additional documentation comments have been added in various places to make the
      logic in the code clearer.
      
      (closes issue #18951)
      Reported by: jmls
      Patches:
            issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
10-digiumphones
Matthew Nicholson 14 years ago
parent 696c77c59e
commit e87639fc26

@ -7438,6 +7438,21 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr,
return p;
}
/*!
* \brief Process the Via header according to RFC 3261 section 18.2.2.
* \param p a sip_pvt structure that will be modified according to the received
* header
* \param req a sip request with a Via header to process
*
* This function will update the destination of the response according to the
* Via header in the request and RFC 3261 section 18.2.2. We do not have a
* transport layer so we ignore certain values like the 'received' param (we
* set the destination address to the addres the request came from in the
* respprep() function).
*
* \retval -1 error
* \retval 0 success
*/
static int process_via(struct sip_pvt *p, const struct sip_request *req)
{
struct sip_via *via = parse_via(get_header(req, "Via"));
@ -7455,17 +7470,13 @@ static int process_via(struct sip_pvt *p, const struct sip_request *req)
return -1;
}
if (via->port) {
ast_sockaddr_set_port(&p->sa, via->port);
} else {
ast_sockaddr_set_port(&p->sa, STANDARD_SIP_PORT);
}
if (ast_sockaddr_is_ipv4_multicast(&p->sa)) {
setsockopt(sipsock, IPPROTO_IP, IP_MULTICAST_TTL, &via->ttl, sizeof(via->ttl));
}
}
ast_sockaddr_set_port(&p->sa, via->port ? via->port : STANDARD_SIP_PORT);
free_via(via);
return 0;
}
@ -10012,6 +10023,9 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg
/* default to routing the response to the address where the request
* came from. Since we don't have a transport layer, we do this here.
* The process_via() function will update the port to either the port
* specified in the via header or the default port later on (per RFC
* 3261 section 18.2.2).
*/
p->sa = p->recv;

@ -2253,10 +2253,7 @@ void free_via(struct sip_via *v)
return;
}
if (v->via) {
ast_free(v->via);
}
ast_free(v->via);
ast_free(v);
}

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