add manual_rtp_bug gen_one_gen_all to prevent rtp passthru to break the *S* word..

diff --git a/src/include/switch_types.h b/src/include/switch_types.h
index 21dfa513395f2e74205b3a36a60839dd3f0dadf5..bcdc508408613f12418c17044117e15e7e5c1f3c 100644 (file)
--- a/src/include/switch_types.h
+++ b/src/include/switch_types.h
@@ -685,7 +685,7 @@ typedef enum {
         */
 
 
-       RTP_BUG_ACCEPT_ANY_PACKETS = (1 << 7)
+       RTP_BUG_ACCEPT_ANY_PACKETS = (1 << 7),
 
        /*
          Oracle's Contact Center Anywhere (CCA) likes to use a single RTP socket to send all its outbound audio.
@@ -694,6 +694,21 @@ typedef enum {
        */
 
 
+       RTP_BUG_GEN_ONE_GEN_ALL = (1 << 8)
+
+       /*
+         Some RTP endpoints (and by some we mean *cough* _SONUS_!) do not like it when the timestamps jump forward or backwards in time.
+         So say you are generating a file that says "please wait for me to complete your call, or generating ringback"
+         Now you place and outbound call and you are bridging.  Well, while you were playing the file, you were generating your own RTP timestamps.
+         But, now that you have a remote RTP stream, you'd rather send those timestamps as-is in case they will be fed to a remote jitter buffer......
+         Ok, so this causes the audio to completely fade out despite the fact that we send the mark bit which should give them heads up its happening.
+
+         Sigh, This flag will tell FreeSWITCH that if it ever generates even one RTP packet itself, to continue to generate all of them and ignore the
+         actual timestamps in the frames.
+
+        */
+
+
 } switch_rtp_bug_flag_t;
 
 #ifdef _MSC_VER

diff --git a/src/mod/applications/mod_dptools/mod_dptools.c b/src/mod/applications/mod_dptools/mod_dptools.c
index fe82187d6c6d0a2abb2165105aa2b5c40524ecbc..857a0ea968b47afd9ed0683b40124d0480332cfa 100755 (executable)
--- a/src/mod/applications/mod_dptools/mod_dptools.c
+++ b/src/mod/applications/mod_dptools/mod_dptools.c
@@ -3730,8 +3730,8 @@ static int next_file(switch_file_handle_t *handle)
        }
 
        handle->samples = context->fh.samples;
-       handle->samplerate = context->fh.samplerate;
-       handle->channels = context->fh.channels;
+       //handle->samplerate = context->fh.samplerate;
+       //handle->channels = context->fh.channels;
        handle->format = context->fh.format;
        handle->sections = context->fh.sections;
        handle->seekable = context->fh.seekable;

diff --git a/src/mod/endpoints/mod_sofia/sofia_glue.c b/src/mod/endpoints/mod_sofia/sofia_glue.c
index 518f1065b81868580c8a4cff411028189df378c9..93414e29f78dacecc2411002ad754b010e918005 100644 (file)
--- a/src/mod/endpoints/mod_sofia/sofia_glue.c
+++ b/src/mod/endpoints/mod_sofia/sofia_glue.c
@@ -6577,6 +6577,14 @@ void sofia_glue_parse_rtp_bugs(switch_rtp_bug_flag_t *flag_pole, const char *str
        if (switch_stristr("~ACCEPT_ANY_PACKETS", str)) {
                *flag_pole &= ~RTP_BUG_ACCEPT_ANY_PACKETS;
        }
+
+       if (switch_stristr("GEN_ONE_GEN_ALL", str)) {
+               *flag_pole |= RTP_BUG_GEN_ONE_GEN_ALL;
+       }
+
+       if (switch_stristr("~GEN_ONE_GEN_ALL", str)) {
+               *flag_pole &= ~RTP_BUG_GEN_ONE_GEN_ALL;
+       }
 }
 
 char *sofia_glue_gen_contact_str(sofia_profile_t *profile, sip_t const *sip, sofia_dispatch_event_t *de, sofia_nat_parse_t *np)

diff --git a/src/switch_rtp.c b/src/switch_rtp.c
index c8a5777c3e3d5ec09ea8eed6088fc46236458a5b..0ea3a978b049c84e415b554f3f50b5e30dfb5cc8 100644 (file)
--- a/src/switch_rtp.c
+++ b/src/switch_rtp.c
@@ -4175,6 +4175,11 @@ SWITCH_DECLARE(int) switch_rtp_write_frame(switch_rtp_t *rtp_session, switch_fra
 
        fwd = (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE) && switch_test_flag(frame, SFF_RAW_RTP)) ? 1 : 0;
 
+       if (!fwd && switch_test_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE) && (rtp_session->rtp_bugs & RTP_BUG_GEN_ONE_GEN_ALL)) {
+               switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Generating RTP locally but timestamp passthru is configured, disabling....\n");
+               switch_clear_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE);
+       }
+
        switch_assert(frame != NULL);
 
        if (switch_test_flag(frame, SFF_CNG)) {