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Diffstat (limited to 'channels/chan_pjsip.c')
-rw-r--r--channels/chan_pjsip.c108
1 files changed, 81 insertions, 27 deletions
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 6eea79368..75dcd7610 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -727,27 +727,45 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
session = channel->session;
- if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
- ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
- ast_format_get_name(f->subclass.format), ast_channel_name(ast));
-
- ast_frfree(f);
- return &ast_null_frame;
- }
-
+ /*
+ * Asymmetric RTP only has one native format set at a time.
+ * Therefore we need to update the native format to the current
+ * raw read format BEFORE the native format check
+ */
if (!session->endpoint->asymmetric_rtp_codec &&
ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
- /* For maximum compatibility we ensure that the write format matches that of the received media */
+ struct ast_format_cap *caps;
+
+ /* For maximum compatibility we ensure that the formats match that of the received media */
ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
ast_format_get_name(f->subclass.format), ast_channel_name(ast),
ast_format_get_name(ast_channel_rawwriteformat(ast)));
+
+ caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (caps) {
+ ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
+ ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
+ ast_format_cap_append(caps, f->subclass.format, 0);
+ ast_channel_nativeformats_set(ast, caps);
+ ao2_ref(caps, -1);
+ }
+
ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format);
+ ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format);
if (ast_channel_is_bridged(ast)) {
ast_channel_set_unbridged_nolock(ast, 1);
}
}
+ if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
+ ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
+ ast_format_get_name(f->subclass.format), ast_channel_name(ast));
+
+ ast_frfree(f);
+ return &ast_null_frame;
+ }
+
if (session->dsp) {
int dsp_features;
@@ -827,6 +845,8 @@ static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
break;
case AST_FRAME_MODEM:
break;
+ case AST_FRAME_CNG:
+ break;
default:
ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
break;
@@ -1259,8 +1279,7 @@ static int update_connected_line_information(void *data)
int generate_new_sdp;
method = session->endpoint->id.refresh_method;
- if (session->inv_session->invite_tsx
- && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
+ if (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE) {
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
}
@@ -1364,7 +1383,8 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
/* FIXME: Only use this for VP8. Additional work would have to be done to
* fully support other video codecs */
- if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL) {
+ if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL ||
+ ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL) {
/* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
* RTP engine would provide a way to externally write/schedule RTCP
* packets */
@@ -1688,21 +1708,27 @@ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
- switch (channel->session->endpoint->dtmf) {
+ switch (channel->session->dtmf) {
case AST_SIP_DTMF_RFC_4733:
if (!media || !media->rtp) {
return -1;
}
ast_rtp_instance_dtmf_begin(media->rtp, digit);
- break;
+ break;
case AST_SIP_DTMF_AUTO:
- if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
- return -1;
- }
+ if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
+ return -1;
+ }
- ast_rtp_instance_dtmf_begin(media->rtp, digit);
- break;
+ ast_rtp_instance_dtmf_begin(media->rtp, digit);
+ break;
+ case AST_SIP_DTMF_AUTO_INFO:
+ if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
+ return -1;
+ }
+ ast_rtp_instance_dtmf_begin(media->rtp, digit);
+ break;
case AST_SIP_DTMF_NONE:
break;
case AST_SIP_DTMF_INBAND:
@@ -1802,7 +1828,21 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
- switch (channel->session->endpoint->dtmf) {
+ switch (channel->session->dtmf) {
+ case AST_SIP_DTMF_AUTO_INFO:
+ {
+ if (!media || !media->rtp) {
+ return -1;
+ }
+ if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {
+ ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
+ ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
+ break;
+ }
+ /* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
+ ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 NOT negotiated using INFO instead.\n", ast_channel_name(ast));
+ }
+
case AST_SIP_DTMF_INFO:
{
struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
@@ -1835,14 +1875,15 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
}
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
- break;
- case AST_SIP_DTMF_AUTO:
- if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
- return -1;
- }
+ break;
+ case AST_SIP_DTMF_AUTO:
+ if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
+ return -1;
+ }
+
+ ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
+ break;
- ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
- break;
case AST_SIP_DTMF_NONE:
break;
@@ -2599,6 +2640,12 @@ static struct ast_custom_function media_offer_function = {
.write = pjsip_acf_media_offer_write
};
+static struct ast_custom_function dtmf_mode_function = {
+ .name = "PJSIP_DTMF_MODE",
+ .read = pjsip_acf_dtmf_mode_read,
+ .write = pjsip_acf_dtmf_mode_write
+};
+
static struct ast_custom_function session_refresh_function = {
.name = "PJSIP_SEND_SESSION_REFRESH",
.write = pjsip_acf_session_refresh_write,
@@ -2643,6 +2690,11 @@ static int load_module(void)
goto end;
}
+ if (ast_custom_function_register(&dtmf_mode_function)) {
+ ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
+ goto end;
+ }
+
if (ast_custom_function_register(&session_refresh_function)) {
ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
goto end;
@@ -2702,6 +2754,7 @@ static int load_module(void)
end:
ao2_cleanup(pjsip_uids_onhold);
pjsip_uids_onhold = NULL;
+ ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&session_refresh_function);
@@ -2724,6 +2777,7 @@ static int unload_module(void)
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
+ ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&session_refresh_function);