# 27_pjmedia_rebind_remote_peer.patch   (rebased for pjsip 2.17)
#
# Add an in-place "rebind remote peer" helper to pjmedia, used by
# python-sipsimple's AudioStream.update() to handle a port-change
# re-INVITE without going through media_stop / media_start.
#
# Rebase notes (vs the 2.12 version):
#   * Depends on 20_sylk_aead_transport.patch having been applied
#     first (creates sylk_aead_transport.c with the transport_attach2
#     hook this patch hardens).
#   * pjsip 2.17 transport.h uses 4-space indent for the
#     pjmedia_transport_attach2 wrapper (matches 2.12 — the
#     2.13 coding-style sweep was a no-op here, the wrapper was
#     already space-indented).
#   * Trailing-whitespace cleanup hunk on the existing attach2
#     wrapper body is dropped — 2.17 already has the relaxed form.
#
# Behaviour (unchanged from the 2.12 header):
#   1. Adds inline pjmedia_transport_rebind_remote_peer() in
#      pjmedia/include/pjmedia/transport.h. Builds a
#      pjmedia_transport_attach_param with new rem_addr/rem_rtcp
#      and NULL callbacks, then calls pjmedia_transport_attach2.
#      transport_udp::transport_attach() overwrites rem_rtp_addr /
#      rem_rtcp_addr in place; transport_srtp::transport_attach2()
#      skips callback rebinding when no callbacks are passed in.
#      SRTP keys, ROC counters, ssrc state and AEAD wrapper state
#      all survive.
#   2. Hardens sylk_aead_transport.c::transport_attach2 to tolerate
#      a re-attach (NULL callbacks = address-rebind-only; real
#      callbacks = deliberate stream rebind, still works).
#
# python-sipsimple's RTPTransport.rebind_remote_peer() in
# sipsimple/core/_core.mediatransport.pxi and AudioStream.update()
# in sipsimple/streams/rtp/audio.py call into this; those changes
# live outside deps/patches/.
#
--- pjsip_orig/pjmedia/include/pjmedia/transport.h
+++ pjsip/pjmedia/include/pjmedia/transport.h
@@ -772,6 +772,36 @@
 }


+/**
+ * Rebind the remote RTP/RTCP peer addresses of an already-attached
+ * transport, in place, without going through media_stop/media_start.
+ * Used by python-sipsimple's AudioStream.update() for port-change
+ * re-INVITEs — SRTP keys, ROC counters and any AEAD wrapper state
+ * survive the address swap. See 27_pjmedia_rebind_remote_peer.patch.
+ */
+PJ_INLINE(pj_status_t) pjmedia_transport_rebind_remote_peer(
+                                  pjmedia_transport *tp,
+                                  const pj_sockaddr_t *rem_addr,
+                                  const pj_sockaddr_t *rem_rtcp,
+                                  unsigned addr_len)
+{
+    pjmedia_transport_attach_param param;
+    pj_bzero(&param, sizeof(param));
+    pj_sockaddr_cp(&param.rem_addr, rem_addr);
+    if (rem_rtcp && pj_sockaddr_has_addr((pj_sockaddr*)rem_rtcp)) {
+        pj_sockaddr_cp(&param.rem_rtcp, rem_rtcp);
+    } else {
+        pj_sockaddr_cp(&param.rem_rtcp, rem_addr);
+        pj_sockaddr_set_port(&param.rem_rtcp,
+            (pj_uint16_t)(pj_sockaddr_get_port((pj_sockaddr*)rem_addr) + 1));
+    }
+    param.addr_len = addr_len;
+    /* user_data and all callbacks intentionally NULL so srtp/aead
+     * attach2 propagate addresses only. */
+    return pjmedia_transport_attach2(tp, &param);
+}
+
+
 /**
  * Attach callbacks to be called on receipt of incoming RTP/RTCP packets.
  * This is just a simple wrapper which calls <tt>attach()</tt> member of
--- pjsip_orig/pjmedia/src/pjmedia/sylk_aead_transport.c
+++ pjsip/pjmedia/src/pjmedia/sylk_aead_transport.c
@@ -575,28 +575,35 @@
     struct sylk_aead_adapter *a = (struct sylk_aead_adapter *)tp;
     pj_status_t status;

-    pj_assert(a->stream_user_data == NULL);
-    a->stream_user_data = att_param->user_data;
-    if (att_param->rtp_cb2) {
-        a->stream_rtp_cb2 = att_param->rtp_cb2;
-    } else {
-        a->stream_rtp_cb = att_param->rtp_cb;
+    /* Tolerate re-attach (see pjmedia_transport_rebind_remote_peer
+     * in pjmedia/include/pjmedia/transport.h). When no callbacks are
+     * supplied, leave captured stream state alone and just propagate
+     * the new addresses down to the slave. */
+    if (att_param->rtp_cb || att_param->rtp_cb2 || att_param->rtcp_cb) {
+        a->stream_user_data = att_param->user_data;
+        if (att_param->rtp_cb2) {
+            a->stream_rtp_cb2 = att_param->rtp_cb2;
+            a->stream_rtp_cb  = NULL;
+        } else {
+            a->stream_rtp_cb  = att_param->rtp_cb;
+            a->stream_rtp_cb2 = NULL;
+        }
+        a->stream_rtcp_cb = att_param->rtcp_cb;
+        a->stream_ref     = att_param->stream;
     }
-    a->stream_rtcp_cb = att_param->rtcp_cb;
-    a->stream_ref = att_param->stream;

-    att_param->rtp_cb2  = &slave_rtp_cb2;
-    att_param->rtp_cb   = NULL;
-    att_param->rtcp_cb  = &slave_rtcp_cb;
+    att_param->rtp_cb2   = &slave_rtp_cb2;
+    att_param->rtp_cb    = NULL;
+    att_param->rtcp_cb   = &slave_rtcp_cb;
     att_param->user_data = a;

     status = pjmedia_transport_attach2(a->slave_tp, att_param);
-    if (status != PJ_SUCCESS) {
+    if (status != PJ_SUCCESS && a->stream_user_data == att_param->user_data) {
         a->stream_user_data = NULL;
-        a->stream_rtp_cb   = NULL;
-        a->stream_rtp_cb2  = NULL;
-        a->stream_rtcp_cb  = NULL;
-        a->stream_ref      = NULL;
+        a->stream_rtp_cb    = NULL;
+        a->stream_rtp_cb2   = NULL;
+        a->stream_rtcp_cb   = NULL;
+        a->stream_ref       = NULL;
     }
     return status;
 }