Merged revisions 197606 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

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  r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
  
  Recorded merge of revisions 197588 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
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    r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
    
    Allow for media to arrive from an alternate source when responding to a reinvite with 491.
    
    When we receive a SIP reinvite, it is possible that we may not be able to process the
    reinvite immediately since we have also sent a reinvite out ourselves. The problem is
    that whoever sent us the reinvite may have also sent a reinvite out to another party,
    and that reinvite may have succeeded.
    
    As a result, even though we are not going to accept the reinvite we just received, it
    is important for us to not have problems if we suddenly start receiving RTP from a new
    source. The fix for this is to grab the media source information from the SDP of the
    reinvite that we receive. This information is passed to the RTP layer so that it will
    know about the alternate source for media.
    
    Review: https://reviewboard.asterisk.org/r/252
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197615 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 17 insertions, 0 deletions

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 5a857f016..98b6b1ffc 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -133,6 +133,23 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
 
 void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
 
+/*! 
+ * \since 1.4.26
+ * \brief set potential alternate source for RTP media
+ *
+ * This function may be used to give the RTP stack a hint that there is a potential
+ * second source of media. One case where this is used is when the SIP stack receives
+ * a REINVITE to which it will be replying with a 491. In such a scenario, the IP and
+ * port information in the SDP of that REINVITE lets us know that we may receive media
+ * from that source/those sources even though the SIP transaction was unable to be completed
+ * successfully
+ *
+ * \param rtp The RTP structure we wish to set up an alternate host/port on
+ * \param alt The address information for the alternate media source
+ * \retval void
+ */
+void ast_rtp_set_alt_peer(struct ast_rtp *rtp, struct sockaddr_in *alt);
+
 /* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
 int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);