-
Notifications
You must be signed in to change notification settings - Fork 2.5k
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
Add quirk for RTSP servers #2909
Merged
Merged
Changes from all commits
Commits
Show all changes
2 commits
Select commit
Hold shift + click to select a range
File filter
Filter by extension
Conversations
Failed to load comments.
Loading
Jump to
Jump to file
Failed to load files.
Loading
Diff view
Diff view
There are no files selected for viewing
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
Original file line number | Diff line number | Diff line change |
---|---|---|
|
@@ -144,6 +144,8 @@ The following options are only valid for the 'rtsp' type: | |
url = RTSP stream URL | ||
rtsp_user = RTSP authorization username, if needed | ||
rtsp_pwd = RTSP authorization password, if needed | ||
rtsp_quirk = Some RTSP servers offer the stream using only the path, instead of the fully qualified URL. | ||
If set true, this boolean informs Janus that we should try a path-only DESCRIBE request if the initial request returns 404. | ||
rtsp_failcheck = whether an error should be returned if connecting to the RTSP server fails (default=true) | ||
rtspiface = network interface IP address or device name to listen on when receiving RTSP streams | ||
rtsp_reconnect_delay = after n seconds passed and no media assumed, the RTSP server has gone and schedule a reconnect (default=5s) | ||
|
@@ -1096,6 +1098,8 @@ typedef struct janus_streaming_rtp_source { | |
janus_streaming_buffer *curldata; | ||
char *rtsp_url; | ||
char *rtsp_username, *rtsp_password; | ||
char *rtsp_stream_uri; | ||
gboolean rtsp_quirk; | ||
gint64 ka_timeout; | ||
char *rtsp_ahost, *rtsp_vhost; | ||
gboolean reconnecting; | ||
|
@@ -1221,7 +1225,7 @@ janus_streaming_mountpoint *janus_streaming_create_file_source( | |
janus_streaming_mountpoint *janus_streaming_create_rtsp_source( | ||
uint64_t id, char *id_str, char *name, char *desc, char *metadata, | ||
char *url, char *username, char *password, | ||
gboolean doaudio, int audiopt, char *artpmap, char *afmtp, | ||
gboolean quirk, gboolean doaudio, int audiopt, char *artpmap, char *afmtp, | ||
gboolean dovideo, int videopt, char *vrtpmap, char *vfmtp, gboolean bufferkf, | ||
const janus_network_address *iface, int threads, | ||
gint64 reconnect_delay, gint64 session_timeout, int rtsp_timeout, int rtsp_conn_timeout, | ||
|
@@ -2132,6 +2136,7 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) { | |
janus_config_item *file = janus_config_get(config, cat, janus_config_type_item, "url"); | ||
janus_config_item *username = janus_config_get(config, cat, janus_config_type_item, "rtsp_user"); | ||
janus_config_item *password = janus_config_get(config, cat, janus_config_type_item, "rtsp_pwd"); | ||
janus_config_item *quirk = janus_config_get(config, cat, janus_config_type_item, "rtsp_quirk"); | ||
janus_config_item *audio = janus_config_get(config, cat, janus_config_type_item, "audio"); | ||
janus_config_item *artpmap = janus_config_get(config, cat, janus_config_type_item, "audiortpmap"); | ||
janus_config_item *acodec = janus_config_get(config, cat, janus_config_type_item, "audiopt"); | ||
|
@@ -2155,6 +2160,7 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) { | |
continue; | ||
} | ||
gboolean is_private = priv && priv->value && janus_is_true(priv->value); | ||
gboolean rtsp_quirk = quirk && quirk->value && janus_is_true(quirk->value); | ||
gboolean doaudio = audio && audio->value && janus_is_true(audio->value); | ||
gboolean dovideo = video && video->value && janus_is_true(video->value); | ||
gboolean bufferkf = video && vkf && vkf->value && janus_is_true(vkf->value); | ||
|
@@ -2189,6 +2195,7 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) { | |
(char *)file->value, | ||
username ? (char *)username->value : NULL, | ||
password ? (char *)password->value : NULL, | ||
rtsp_quirk, | ||
doaudio, | ||
(acodec && acodec->value) ? atoi(acodec->value) : -1, | ||
artpmap ? (char *)artpmap->value : NULL, | ||
|
@@ -2616,6 +2623,8 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi | |
json_object_set_new(ml, "rtsp_user", json_string(source->rtsp_username)); | ||
if(source->rtsp_password) | ||
json_object_set_new(ml, "rtsp_pwd", json_string(source->rtsp_password)); | ||
if(source->rtsp_quirk) | ||
json_object_set_new(ml, "rtsp_quirk", json_true()); | ||
} | ||
} | ||
#endif | ||
|
@@ -3220,6 +3229,7 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi | |
json_t *url = json_object_get(root, "url"); | ||
json_t *username = json_object_get(root, "rtsp_user"); | ||
json_t *password = json_object_get(root, "rtsp_pwd"); | ||
json_t *quirk = json_object_get(root, "rtsp_quirk"); | ||
json_t *iface = json_object_get(root, "rtspiface"); | ||
json_t *threads = json_object_get(root, "threads"); | ||
json_t *failerr = json_object_get(root, "rtsp_failcheck"); | ||
|
@@ -3231,6 +3241,7 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi | |
failerr = json_object_get(root, "rtsp_check"); | ||
gboolean doaudio = audio ? json_is_true(audio) : FALSE; | ||
gboolean dovideo = video ? json_is_true(video) : FALSE; | ||
gboolean doquirk = quirk ? json_is_true(quirk) : FALSE; | ||
gboolean error_on_failure = failerr ? json_is_true(failerr) : TRUE; | ||
if(!doaudio && !dovideo) { | ||
JANUS_LOG(LOG_ERR, "Can't add 'rtsp' stream, no audio or video have to be streamed...\n"); | ||
|
@@ -3264,6 +3275,7 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi | |
(char *)json_string_value(url), | ||
username ? (char *)json_string_value(username) : NULL, | ||
password ? (char *)json_string_value(password) : NULL, | ||
doquirk, | ||
doaudio, (audiopt ? json_integer_value(audiopt) : -1), | ||
(char *)json_string_value(audiortpmap), (char *)json_string_value(audiofmtp), | ||
dovideo, (videopt ? json_integer_value(videopt) : -1), | ||
|
@@ -3417,6 +3429,8 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi | |
janus_config_add(config, c, janus_config_item_create("rtsp_user", source->rtsp_username)); | ||
if(source->rtsp_password) | ||
janus_config_add(config, c, janus_config_item_create("rtsp_pwd", source->rtsp_password)); | ||
if(source->rtsp_password) | ||
There was a problem hiding this comment. Choose a reason for hiding this commentThe reason will be displayed to describe this comment to others. Learn more. Just noticed a typo: this should be |
||
janus_config_add(config, c, janus_config_item_create("rtsp_quirk", "yes")); | ||
#endif | ||
janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no")); | ||
if(mp->codecs.audio_pt >= 0) { | ||
|
@@ -3631,6 +3645,8 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi | |
janus_config_add(config, c, janus_config_item_create("rtsp_user", source->rtsp_username)); | ||
if(source->rtsp_password) | ||
janus_config_add(config, c, janus_config_item_create("rtsp_pwd", source->rtsp_password)); | ||
if(source->rtsp_password) | ||
There was a problem hiding this comment. Choose a reason for hiding this commentThe reason will be displayed to describe this comment to others. Learn more. Same here. |
||
janus_config_add(config, c, janus_config_item_create("rtsp_quirk", "yes")); | ||
#endif | ||
janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no")); | ||
if(mp->codecs.audio_pt >= 0) { | ||
|
@@ -5795,6 +5811,7 @@ static void janus_streaming_rtp_source_free(janus_streaming_rtp_source *source) | |
g_free(source->rtsp_url); | ||
g_free(source->rtsp_username); | ||
g_free(source->rtsp_password); | ||
g_free(source->rtsp_stream_uri); | ||
g_free(source->rtsp_ahost); | ||
g_free(source->rtsp_vhost); | ||
janus_mutex_unlock(&source->rtsp_mutex); | ||
|
@@ -6519,6 +6536,25 @@ static int janus_streaming_rtsp_connect_to_server(janus_streaming_mountpoint *mp | |
} | ||
long code = 0; | ||
res = curl_easy_getinfo(curl, CURLINFO_RESPONSE_CODE, &code); | ||
#if CURL_AT_LEAST_VERSION(7, 62, 0) | ||
if(source->rtsp_quirk && code == 404) { | ||
/* Possibly a quirk in the RTSP server, where the DESCRIBE request expects a path only. */ | ||
CURLU *curl_u = curl_url(); | ||
char *path = NULL; | ||
if(!(curl_url_set(curl_u, CURLUPART_URL, source->rtsp_url, 0))) { | ||
if(!(curl_url_get(curl_u, CURLUPART_PATH, &path, 0))) { | ||
curl_easy_setopt(curl, CURLOPT_RTSP_STREAM_URI, path); | ||
curl_easy_perform(curl); | ||
res = curl_easy_getinfo(curl, CURLINFO_RESPONSE_CODE, &code); | ||
if((res == CURLE_OK) && (code != 404)) { | ||
source->rtsp_stream_uri = g_strdup(path); | ||
} | ||
curl_free(path); | ||
} | ||
} | ||
curl_url_cleanup(curl_u); | ||
} | ||
#endif | ||
if(res != CURLE_OK) { | ||
JANUS_LOG(LOG_ERR, "Couldn't get DESCRIBE answer: %s\n", curl_easy_strerror(res)); | ||
curl_easy_cleanup(curl); | ||
|
@@ -7073,7 +7109,7 @@ static int janus_streaming_rtsp_play(janus_streaming_rtp_source *source) { | |
source->curldata->buffer = g_malloc0(1); | ||
source->curldata->size = 0; | ||
JANUS_LOG(LOG_VERB, "Sending PLAY request...\n"); | ||
curl_easy_setopt(source->curl, CURLOPT_RTSP_STREAM_URI, source->rtsp_url); | ||
curl_easy_setopt(source->curl, CURLOPT_RTSP_STREAM_URI, source->rtsp_stream_uri ? source->rtsp_stream_uri : source->rtsp_url); | ||
curl_easy_setopt(source->curl, CURLOPT_RANGE, "npt=0.000-"); | ||
curl_easy_setopt(source->curl, CURLOPT_RTSP_REQUEST, (long)CURL_RTSPREQ_PLAY); | ||
int res = curl_easy_perform(source->curl); | ||
|
@@ -7091,7 +7127,7 @@ static int janus_streaming_rtsp_play(janus_streaming_rtp_source *source) { | |
janus_streaming_mountpoint *janus_streaming_create_rtsp_source( | ||
uint64_t id, char *id_str, char *name, char *desc, char *metadata, | ||
char *url, char *username, char *password, | ||
gboolean doaudio, int acodec, char *artpmap, char *afmtp, | ||
gboolean quirk, gboolean doaudio, int acodec, char *artpmap, char *afmtp, | ||
gboolean dovideo, int vcodec, char *vrtpmap, char *vfmtp, gboolean bufferkf, | ||
const janus_network_address *iface, int threads, | ||
gint64 reconnect_delay, gint64 session_timeout, int rtsp_timeout, int rtsp_conn_timeout, | ||
|
@@ -7166,6 +7202,8 @@ janus_streaming_mountpoint *janus_streaming_create_rtsp_source( | |
live_rtsp_source->rtsp_url = g_strdup(url); | ||
live_rtsp_source->rtsp_username = username ? g_strdup(username) : NULL; | ||
live_rtsp_source->rtsp_password = password ? g_strdup(password) : NULL; | ||
live_rtsp_source->rtsp_stream_uri = NULL; | ||
live_rtsp_source->rtsp_quirk = quirk; | ||
live_rtsp_source->arc = NULL; | ||
live_rtsp_source->vrc = NULL; | ||
live_rtsp_source->drc = NULL; | ||
|
@@ -7293,7 +7331,7 @@ janus_streaming_mountpoint *janus_streaming_create_rtsp_source( | |
janus_streaming_mountpoint *janus_streaming_create_rtsp_source( | ||
uint64_t id, char *id_str, char *name, char *desc, char *metadata, | ||
char *url, char *username, char *password, | ||
gboolean doaudio, int acodec, char *audiortpmap, char *audiofmtp, | ||
gboolean quirk, gboolean doaudio, int acodec, char *audiortpmap, char *audiofmtp, | ||
gboolean dovideo, int vcodec, char *videortpmap, char *videofmtp, gboolean bufferkf, | ||
const janus_network_address *iface, int threads, | ||
gint64 reconnect_delay, gint64 session_timeout, int rtsp_timeout, int rtsp_conn_timeout, | ||
|
Add this suggestion to a batch that can be applied as a single commit.
This suggestion is invalid because no changes were made to the code.
Suggestions cannot be applied while the pull request is closed.
Suggestions cannot be applied while viewing a subset of changes.
Only one suggestion per line can be applied in a batch.
Add this suggestion to a batch that can be applied as a single commit.
Applying suggestions on deleted lines is not supported.
You must change the existing code in this line in order to create a valid suggestion.
Outdated suggestions cannot be applied.
This suggestion has been applied or marked resolved.
Suggestions cannot be applied from pending reviews.
Suggestions cannot be applied on multi-line comments.
Suggestions cannot be applied while the pull request is queued to merge.
Suggestion cannot be applied right now. Please check back later.
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
These two lines should be added to
janus.plugin.streaming.jcfg.in
as well.