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rtp_media_server: typos

Дилян Палаузов 2 年之前
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f6c5638e3e

+ 2 - 2
src/modules/rtp_media_server/doc/rtp_media_server_admin.xml

@@ -69,7 +69,7 @@
 				<emphasis>oRTP</emphasis> git clone <ulink url="git://git.linphone.org/ortp.git">git://git.linphone.org/ortp.git</ulink>
 				<emphasis>oRTP</emphasis> git clone <ulink url="git://git.linphone.org/ortp.git">git://git.linphone.org/ortp.git</ulink>
 			</para>
 			</para>
 			<para>
 			<para>
-				oRTP is a library implemeting Real-time Transport Protocol (RFC 3550), distributed under GNU GPLv2 or proprietary license.
+				oRTP is a library implementing Real-time Transport Protocol (RFC 3550), distributed under GNU GPLv2 or proprietary license.
 			</para>
 			</para>
 			</listitem>
 			</listitem>
 			<listitem>
 			<listitem>
@@ -263,7 +263,7 @@ route {
 	<section id="rtp_media_server.f.rms_play">
 	<section id="rtp_media_server.f.rms_play">
 		<title><varname>rms_play</varname> (file, event_route)</title>
 		<title><varname>rms_play</varname> (file, event_route)</title>
 		<para>
 		<para>
-		Play a wav file, a resampler is automaticaly configured to resample
+		Play a wav file, a resampler is automatically configured to resample
 		and convert stereo to mono if needed.
 		and convert stereo to mono if needed.
 		</para><para>
 		</para><para>
 		The second parameter is the event route that will be called when the file was played.
 		The second parameter is the event route that will be called when the file was played.

+ 1 - 1
src/modules/rtp_media_server/rms_sdp.c

@@ -269,7 +269,7 @@ PayloadType *rms_sdp_select_payload(rms_sdp_info_t *sdp)
 		payload_type_number = strtok(NULL, " ");
 		payload_type_number = strtok(NULL, " ");
 	}
 	}
 	if(!pt->mime_type) {
 	if(!pt->mime_type) {
-		LM_INFO("unsuported codec\n");
+		LM_INFO("unsupported codec\n");
 		shm_free(pt); // payload_type_destroy(pt);
 		shm_free(pt); // payload_type_destroy(pt);
 		return NULL;
 		return NULL;
 	}
 	}

+ 3 - 3
src/modules/rtp_media_server/rtp_media_server.c

@@ -209,7 +209,7 @@ static int mod_init(void)
 }
 }
 
 
 /**
 /**
- * Called only once when Kamailio is shuting down to clean up module
+ * Called only once when Kamailio is shutting down to clean up module
  * resources.
  * resources.
  */
  */
 static void mod_destroy()
 static void mod_destroy()
@@ -346,7 +346,7 @@ static void rms_dialog_manage_loop()
  * The rank will be o for the main process calling this function,
  * The rank will be o for the main process calling this function,
  * or 1 through n for each listener process. The rank can have a negative
  * or 1 through n for each listener process. The rank can have a negative
  * value if it is a special process calling the child init function.
  * value if it is a special process calling the child init function.
- * Other then the listeners, the rank will equal one of these values:
+ * Other than the listeners, the rank will equal one of these values:
  * PROC_MAIN      0  Main ser process
  * PROC_MAIN      0  Main ser process
  * PROC_TIMER    -1  Timer attendant process
  * PROC_TIMER    -1  Timer attendant process
  * PROC_FIFO     -2  FIFO attendant process
  * PROC_FIFO     -2  FIFO attendant process
@@ -1096,7 +1096,7 @@ static int rms_sip_request_f(struct sip_msg *msg)
 		if(di->bridged_di) { // bridged
 		if(di->bridged_di) { // bridged
 			LM_NOTICE("BYE in brigde mode\n");
 			LM_NOTICE("BYE in brigde mode\n");
 			rms_sip_forward(di, msg, method);
 			rms_sip_forward(di, msg, method);
-		} else { // connected localy
+		} else { // connected locally
 			LM_NOTICE("BYE in local mode\n");
 			LM_NOTICE("BYE in local mode\n");
 			rms_disconnect(msg);
 			rms_disconnect(msg);
 			return 1;
 			return 1;