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Merged revisions 44971 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r44971 | kpfleming | 2006-10-12 14:14:24 -0500 (Thu, 12 Oct 2006) | 2 lines

we can only send one 'a=ptime' attribute per media session, not one for each format

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-10-12 19:15:25 +00:00
parent 035aeb82e4
commit 0ecfae3dbd
1 changed files with 17 additions and 11 deletions
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channels/chan_sip.c
+17 -11
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@@ -1247,7 +1247,7 @@ static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req);
static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug);
int debug, int *min_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug);
@@ -5638,7 +5638,7 @@ static int add_vidupdate(struct sip_request *req)
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug)
int debug, int *min_packet_size)
{
int rtp_code;
struct ast_format_list fmt;
@@ -5667,9 +5667,8 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
}
if (codec != AST_FORMAT_ILBC) {
ast_build_string(a_buf, a_size, "a=ptime:%d\r\n", fmt.cur_ms);
}
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
*min_packet_size = fmt.cur_ms;
}
/*! \brief Get Max T.38 Transmission rate from T38 capabilities */
@@ -5863,6 +5862,8 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
int capability;
int needvideo = FALSE;
int debug = sip_debug_test_pvt(p);
int min_audio_packet_size = 0;
int min_video_packet_size = 0;
m_video[0] = '\0'; /* Reset the video media string if it's not needed */
@@ -5995,11 +5996,10 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
add_codec_to_sdp(p, p->prefcodec & AST_FORMAT_AUDIO_MASK, 8000,
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug);
debug, &min_audio_packet_size);
alreadysent |= p->prefcodec & AST_FORMAT_AUDIO_MASK;
}
/* Start by sending our preferred audio codecs */
for (x = 0; x < 32; x++) {
int pref_codec;
@@ -6016,7 +6016,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
add_codec_to_sdp(p, pref_codec, 8000,
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug);
debug, &min_audio_packet_size);
alreadysent |= pref_codec;
}
@@ -6032,12 +6032,12 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
add_codec_to_sdp(p, x, 8000,
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug);
debug, &min_audio_packet_size);
else
add_codec_to_sdp(p, x, 90000,
&m_video_next, &m_video_left,
&a_video_next, &a_video_left,
debug);
debug, &min_video_packet_size);
}
/* Now add DTMF RFC2833 telephony-event as a codec */
@@ -6054,9 +6054,15 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
if (option_debug > 2)
ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n");
if(!p->owner || !ast_internal_timing_enabled(p->owner))
if (!p->owner || !ast_internal_timing_enabled(p->owner))
ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");
if (min_audio_packet_size)
ast_build_string(&a_audio_next, &a_audio_left, "a=ptime:%d\r\n", min_audio_packet_size);
if (min_video_packet_size)
ast_build_string(&a_video_next, &a_video_left, "a=ptime:%d\r\n", min_video_packet_size);
if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0))
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");