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authorGeorge Joseph <george.joseph@fairview5.com>2016-01-07 10:57:01 -0700
committerGeorge Joseph <george.joseph@fairview5.com>2016-01-11 18:41:31 -0600
commita41aab477ac317f9d973db253b4c30cd6a6db5b8 (patch)
tree7921ac4515db887d05e00dd116cd5b90536d639d /CHANGES
parent188438c53f5c5424cd45c3b904c58508881f0baa (diff)
pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address
On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES8
1 files changed, 8 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index 8d5f5b388..6885c512a 100644
--- a/CHANGES
+++ b/CHANGES
@@ -234,6 +234,14 @@ Voicemail
app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
modules.conf to force app_voicemail to be the voicemail provider.
+res_pjsip_sdp_rtp
+------------------
+ * A new option (bind_rtp_to_media_address) has been added to endpoint which
+ will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
+ media_address as well as using it in the SDP. If set, RTP packets will now
+ originate from the media address instead of the operating system's "primary"
+ ip address.
+
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--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
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