[asterisk-dev] SIP/TDM interworking, and RTP on CALL PROCEEDING

Richard Mudgett rmudgett at digium.com
Mon Feb 18 11:13:38 CST 2013

> >> >I would like to see a "pri set debug on span x" trace of the
> >> >call.
> >>
> >> I am attaching what I hope is enough. The debug is actually
> >> "intense", and I
> >
> >That trace is not intense.  It is decoding the Q.931 (layer 3)
> >messages.
> >The intense trace is *only* needed when debugging Q.921 (layer 2)
> >problems.
> >Otherwise, the intense trace just fills the log with unnecessary
> >noise.
> I'm sorry. I did not intend to do something not in your requests. It
> was not my
> intention to make the log less readable, or make you spend more time
> in reading
> unnecessary details.

You supplied exactly what I asked.  I was just commenting on your
description of the trace as "intense".

> >> overwrote references to actual numbers. In this call, which is
> >> made
> >> to a test
> >> destination, CALL PROCEEDING and further progress informations are
> >> purposely
> >> separated by 5 seconds of wait. The RTP flow seems to start after
> >> the
> >> PRI_EVENT_PROCEEDING gets indicated on the SIP channel, and the
> >> corresponding
> >> 100 Trying gets sent.
> >
> >I think you may be confusing things.  The PRI_EVENT_PROCEEDING is
> >associated with the DAHDI channel not the SIP channel.  Messages
> >from different threads can come out delayed relative to when things
> >actually happen between threads.
> I may, indeed, be confusing something. As I said I surely do not
> match your
> expertise in the field, but I think that I was possibly misleading in
> my
> explanation. What I meant to say, (I hope) more properly is that what
> I see
> is:
>  - CALL PROCEEDING coming on the DAHDI channel
>  - 100 Trying on the SIP channel
>  - RTP being sent immediately after this, and before any other
> progress indication
>    (being it PROGRESS, ALERTING, with relevant progress indication 1
>    or 8)
> >The trace shows Asterisk receiving
> >PROCEEDING (with no progress indication ie)
> >wait 5 seconds
> >PROGRESS (with progress indication ie)
> >wait 5 seconds
> >ALERTING (with progress indication ie)
> >
> >The PROCEEDING message will cause the SIP channel to send a
> >"100 Trying" message.  The SIP message should not have a SDP body.
> >The PROCEEDING message does not have a progress indication ie and
> >will not open the media path from the DAHDI channel.
> >
> >The PROGRESS message will cause the SIP channel to send a
> >"183 Session Progress" message.  There should also be a SDP
> >body to open the RTP media path to the calling phone.  The
> >PROGRESS message *does* have a progress indication ie and *will*
> >open the media path from the DAHDI channel.
> >
> >The ALERTING message may cause the SIP channel to send a
> >"180 Ringing" if it has not previously send a 183 message.
> >The ALERTING message also has a progress indication ie that would
> >open the media path from the DAHDI channel if it were not already
> >opened by the previous PROGRESS message.
> >
> >This behavior would be expected from the messages in the trace.
> I can confirm that your description of what seen on the trace is
> flawless,
> however, nonetheless it seems that the RTP flow starts right after
> the CALL
> PROCEEDING, and before receiving other progress information. As I
> said, this is what
> I was not expecting. I thought (as you do, I think), that a CALL
> PROCEEDING without
> any progress indication 1 or 8 would lead to sending only a 100
> Trying
> on the SIP
> channel, without leading to any media being sent.

Since your trace has a PROGRESS message that *will* open the media
path before the ALERTING message comes in, here is a test change
that will disable posting the PROGRESS control frame to the SIP

Index: channels/sig_pri.c
--- channels/sig_pri.c  (revision 381627)
+++ channels/sig_pri.c  (working copy)
@@ -5617,6 +5617,7 @@

+#if 0  /* For testing purposes ONLY. */
                                if (!pri->pvts[chanpos]->progress
                                        && !pri->pvts[chanpos]->no_b_channel
@@ -5636,6 +5637,7 @@
                                        sig_pri_set_dialing(pri->pvts[chanpos], 0);
+#endif /* For testing purposes ONLY. */
                        case PRI_EVENT_PROCEEDING:

Please try this to see if this suppresses your early RTP stream.


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