[Asterisk-code-review] chan pjsip: Handle T.38 faxes with direct media bridges (asterisk[13])
Matt Jordan
asteriskteam at digium.com
Sun Nov 22 22:35:55 CST 2015
Matt Jordan has uploaded a new change for review.
https://gerrit.asterisk.org/1688
Change subject: chan_pjsip: Handle T.38 faxes with direct media bridges
......................................................................
chan_pjsip: Handle T.38 faxes with direct media bridges
When a channel is in a direct media bridge, a re-INVITE may arrive that forces
Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
must change its technology to a simple bridge, and re-INVITE the media back
to Asterisk.
Generally, this logic mostly already exists in Asterisk. However, prior to this
patch, there were a few bugs:
(1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
ever entering into a direct media bridge. This applies even when the only
media being passed over the channel is audio. This patch fixes this bug
by having the framehook specify that it defers caring about any frame type.
This allows the channels to enter into a direct media bridge, which will
be broken when a re-INVITE is received.
(2) When a re-INVITE is received, nothing instructed the bridging layer to
re-inspect the allowed bridging technology. This now occurs when either
a re-INVITE is received from a peer, or when a response is received from
the far end (that is, when the T.38 state changes to either
T38_PEER_REINVITE or T38_LOCAL_REINVITE).
(3) chan_pjsip needs to do a small amount of work to prevent a direct media
bridge from being chosen when a T.38 session is in progress. When a T.38
session supplement has a t38 datastore - which is added when we detect
we should start thinking about T.38 on a channel - we now refuse a native
RTP bridge.
(4) When a BYE request is received, we don't terminate the T.38 session. If
the other side of a T.38 fax survives the hangup (due to the 'g' flag
in Dial, for example), we don't currently re-INVITE the media on the
other channel back to audio. This patch now has res_pjsip_t38 intercept
BYE requests and inform the far side that the T.38 session is terminated.
This naturally causes the correct re-INVITEs to be sent.
ASTERISK-25582
Change-Id: I7c3575d02b2b6b82ebb6c969cb01545b57a95304
---
M channels/chan_pjsip.c
M res/res_pjsip_t38.c
2 files changed, 59 insertions(+), 1 deletion(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/88/1688/1
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 1319094..0a8d1bc 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -162,11 +162,18 @@
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct chan_pjsip_pvt *pvt;
struct ast_sip_endpoint *endpoint;
+ struct ast_datastore *datastore;
if (!channel || !channel->session || !(pvt = channel->pvt) || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
+ datastore = ast_sip_session_get_datastore(channel->session, "t38");
+ if (datastore) {
+ ao2_ref(datastore, -1);
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
+
endpoint = channel->session->endpoint;
*instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
diff --git a/res/res_pjsip_t38.c b/res/res_pjsip_t38.c
index 0db2e5c..2c544bb 100644
--- a/res/res_pjsip_t38.c
+++ b/res/res_pjsip_t38.c
@@ -161,6 +161,9 @@
parameters.max_ifp = ast_udptl_get_far_max_ifp(session_media->udptl);
parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
ast_udptl_set_tag(session_media->udptl, "%s", ast_channel_name(session->channel));
+
+ /* Inform the bridge the channel is in that it needs to be reconfigured */
+ ast_channel_set_unbridged(session->channel, 1);
break;
case T38_ENABLED:
parameters = state->their_parms;
@@ -177,7 +180,8 @@
}
break;
case T38_LOCAL_REINVITE:
- /* wait until we get a peer response before responding to local reinvite */
+ /* Inform the bridge the channel is in that it needs to be reconfigured */
+ ast_channel_set_unbridged(session->channel, 1);
break;
case T38_MAX_ENUM:
/* Well, that shouldn't happen */
@@ -463,6 +467,11 @@
ast_framehook_detach(new_chan, framehook_id);
}
+static int t38_consume(void *data, enum ast_frame_type type)
+{
+ return 0;
+}
+
static const struct ast_datastore_info t38_framehook_datastore = {
.type = "T38 framehook",
};
@@ -475,6 +484,7 @@
static struct ast_framehook_interface hook = {
.version = AST_FRAMEHOOK_INTERFACE_VERSION,
.event_cb = t38_framehook,
+ .consume_cb = t38_consume,
.chan_fixup_cb = t38_masq,
.chan_breakdown_cb = t38_masq,
};
@@ -558,6 +568,41 @@
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL + 1,
.incoming_request = t38_incoming_invite_request,
.outgoing_request = t38_outgoing_invite_request,
+};
+
+static int t38_incoming_bye_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+ struct ast_datastore *datastore;
+ struct ast_sip_session_media *session_media;
+
+ if (!session->channel) {
+ return 0;
+ }
+
+ datastore = ast_sip_session_get_datastore(session, "t38");
+ if (!datastore) {
+ return 0;
+ }
+
+ session_media = ao2_find(session->media, "image", OBJ_KEY);
+ if (!session_media) {
+ ao2_ref(datastore, -1);
+ return 0;
+ }
+
+ t38_change_state(session, session_media, datastore->data, T38_REJECTED);
+
+ ao2_ref(datastore, -1);
+ ao2_ref(session_media, -1);
+
+ return 0;
+}
+
+/*! \brief Supplement for handling a remote termination of T.38 state */
+static struct ast_sip_session_supplement t38_bye_supplement = {
+ .method = "BYE",
+ .priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL + 1,
+ .incoming_request = t38_incoming_bye_request,
};
/*! \brief Parse a T.38 image stream and store the attribute information */
@@ -889,6 +934,7 @@
{
ast_sip_session_unregister_sdp_handler(&image_sdp_handler, "image");
ast_sip_session_unregister_supplement(&t38_supplement);
+ ast_sip_session_unregister_supplement(&t38_bye_supplement);
return 0;
}
@@ -915,6 +961,11 @@
goto end;
}
+ if (ast_sip_session_register_supplement(&t38_bye_supplement)) {
+ ast_log(LOG_ERROR, "Unable to register T.38 BYE session supplement\n");
+ goto end;
+ }
+
if (ast_sip_session_register_sdp_handler(&image_sdp_handler, "image")) {
ast_log(LOG_ERROR, "Unable to register SDP handler for image stream type\n");
goto end;
--
To view, visit https://gerrit.asterisk.org/1688
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-MessageType: newchange
Gerrit-Change-Id: I7c3575d02b2b6b82ebb6c969cb01545b57a95304
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Matt Jordan <mjordan at digium.com>
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