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authorJoshua Colp <jcolp@digium.com>2015-08-03 08:43:50 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2015-08-03 08:43:50 -0500
commit20ee33e22e3724c35ab3078c5032b87af5b4920c (patch)
tree5ff17b8ac0c7e4fa448934ce312b42d45afd3c45 /include
parente28fbebc575065461964f5aac8246a3e5cdb9a03 (diff)
parent89b21fd9a38bcd89402249440c1670ce48781f30 (diff)
Merge topic 'misc_rtp_tweaks' into 13
* changes: rtp_engine.h: No sense allowing payload types larger than RFC allows. rtp_engine.c: Minor tweaks. rtp_engine.h: Misc comment fixes. chan_sip.c: Tweak glue->update_peer() parameter nil value.
Diffstat (limited to 'include')
-rw-r--r--include/asterisk/rtp_engine.h18
1 files changed, 9 insertions, 9 deletions
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index b7ac2a149..80183ede7 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -1,4 +1,4 @@
- /*
+/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2009, Digium, Inc.
@@ -78,14 +78,13 @@ extern "C" {
#include "asterisk/stasis.h"
#include "asterisk/vector.h"
-/* Maximum number of payloads supported */
-#if defined(LOW_MEMORY)
+/*! Maximum number of payload types RTP can support. */
#define AST_RTP_MAX_PT 128
-#else
-#define AST_RTP_MAX_PT 196
-#endif
-/* Maximum number of generations */
+/*! First dynamic RTP payload type */
+#define AST_RTP_PT_FIRST_DYNAMIC 96
+
+/*! Maximum number of generations */
#define AST_RED_MAX_GENERATION 5
/*!
@@ -625,7 +624,7 @@ struct ast_rtp_glue {
enum ast_rtp_glue_result (*get_trtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
/*! Callback for updating the destination that the remote side should send RTP to */
int (*update_peer)(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active);
- /*! Callback for retrieving codecs that the channel can do. Result returned in result_cap*/
+ /*! Callback for retrieving codecs that the channel can do. Result returned in result_cap. */
void (*get_codec)(struct ast_channel *chan, struct ast_format_cap *result_cap);
/*! Linked list information */
AST_RWLIST_ENTRY(ast_rtp_glue) entry;
@@ -1418,7 +1417,7 @@ unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, struct ast_format
* \code
* struct ast_format_cap *astformats = ast_format_cap_alloc_nolock()
* int nonastformats;
- * ast_rtp_codecs_payload_formats(&codecs, &astformats, &nonastformats);
+ * ast_rtp_codecs_payload_formats(&codecs, astformats, &nonastformats);
* \endcode
*
* This retrieves all the formats known about in the codecs structure and puts the Asterisk ones in the integer
@@ -1449,6 +1448,7 @@ void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_fo
* \since 1.8
*/
int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code);
+
/*!
* \brief Search for a payload code in the ast_rtp_codecs structure
*