Fix #1165:

 - Fixed pjsua_media_channel_create_sdp() to re-calculate audio index of the remote offer, instead of using existing audio index calculated by pjsua_media_channel_init(), as for subsequent SDP offer/answer, pjsua_media_channel_init() may not be called.
 - Fixed SRTP transport to be able to switch SRTP status from active to inactive/by-passed and vice versa.



git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3376 74dad513-b988-da41-8d7b-12977e46ad98

2 files changed, 19 insertions, 1 deletions

diff --git a/pjmedia/src/pjmedia/transport_srtp.c b/pjmedia/src/pjmedia/transport_srtp.c
index 1c6c7f93..4a1d08de 100644
--- a/pjmedia/src/pjmedia/transport_srtp.c
+++ b/pjmedia/src/pjmedia/transport_srtp.c
@@ -645,6 +645,8 @@ PJ_DEF(pj_status_t) pjmedia_transport_srtp_stop(pjmedia_transport *srtp)
     }
 
     p_srtp->session_inited = PJ_FALSE;
+    pj_bzero(&p_srtp->rx_policy, sizeof(p_srtp->rx_policy));
+    pj_bzero(&p_srtp->tx_policy, sizeof(p_srtp->tx_policy));
 
     pj_lock_release(p_srtp->mutex);
 
@@ -1221,6 +1223,7 @@ static pj_status_t transport_encode_sdp(pjmedia_transport *tp,
     PJ_ASSERT_RETURN(tp && sdp_pool && sdp_local, PJ_EINVAL);
     
     srtp->offerer_side = sdp_remote == NULL;
+    srtp->bypass_srtp = PJ_FALSE;
 
     m_rem = sdp_remote ? sdp_remote->media[media_index] : NULL;
     m_loc = sdp_local->media[media_index];
@@ -1576,6 +1579,9 @@ static pj_status_t transport_media_start(pjmedia_transport *tp,
 BYPASS_SRTP:
     srtp->bypass_srtp = PJ_TRUE;
     srtp->peer_use = PJMEDIA_SRTP_DISABLED;
+    if (srtp->session_inited) {
+	pjmedia_transport_srtp_stop(tp);
+    }
 
 PROPAGATE_MEDIA_START:
     return pjmedia_transport_media_start(srtp->member_tp, pool, 
diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c
index 28920079..8ede0f8c 100644
--- a/pjsip/src/pjsua-lib/pjsua_media.c
+++ b/pjsip/src/pjsua-lib/pjsua_media.c
@@ -1247,7 +1247,7 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 	pj_bool_t srtp_active;
 
 #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-	srtp_active = acc->cfg.use_srtp && srtp != NULL;
+	srtp_active = acc->cfg.use_srtp;
 #else
 	srtp_active = PJ_FALSE;
 #endif
@@ -1301,6 +1301,18 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id,
 	return PJ_EBUSY;
     }
 
+    if (rem_sdp && rem_sdp->media_count != 0) {
+	pj_bool_t srtp_active;
+
+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+        srtp_active = pjsua_var.acc[call->acc_id].cfg.use_srtp;
+#else
+	srtp_active = PJ_FALSE;
+#endif
+
+	call->audio_idx = find_audio_index(rem_sdp, srtp_active);
+    }
+
     /* Media index must have been determined before */
     pj_assert(call->audio_idx != -1);