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FS-3172 this also fixes the incorrect usage of L16 on payload 10 which may or may not break interop with other sip devices if we do it right. also added rtp_disable_byteswap variable that can be set to false to disable byteswap when a device is encountered that is incompat (inluding all precious version of FS up till now)

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This commit is contained in:
Anthony Minessale 2011-03-21 14:31:10 -05:00
parent 244fd68dfd
commit e657e32fca
4 changed files with 25 additions and 4 deletions
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2
src/include/switch_types.h
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@ -542,7 +542,7 @@ typedef enum {
SWITCH_RTP_FLAG_BREAK = (1 << 10),
SWITCH_RTP_FLAG_UDPTL = (1 << 11),
SWITCH_RTP_FLAG_DATAWAIT = (1 << 12),
SWITCH_RTP_FLAG_BUGGY_2833 = (1 << 13),
SWITCH_RTP_FLAG_BYTESWAP = (1 << 13),
SWITCH_RTP_FLAG_PASS_RFC2833 = (1 << 14),
SWITCH_RTP_FLAG_AUTO_CNG = (1 << 15),
SWITCH_RTP_FLAG_SECURE_SEND_RESET = (1 << 16),

11
src/mod/endpoints/mod_sofia/sofia_glue.c
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@ -2965,6 +2965,17 @@ switch_status_t sofia_glue_activate_rtp(private_object_t *tech_pvt, switch_rtp_f
flags |= SWITCH_RTP_FLAG_AUTO_CNG;
}
#if __BYTE_ORDER == __LITTLE_ENDIAN
if (!strcasecmp(tech_pvt->read_impl.iananame, "L16")) {
flags |= SWITCH_RTP_FLAG_BYTESWAP;
}
#endif
if ((flags & SWITCH_RTP_FLAG_BYTESWAP) && (val = switch_channel_get_variable(tech_pvt->channel, "rtp_disable_byteswap")) && switch_true(val)) {
flags &= ~SWITCH_RTP_FLAG_BYTESWAP;
}
if (tech_pvt->rtp_session && sofia_test_flag(tech_pvt, TFLAG_REINVITE)) {
//const char *ip = switch_channel_get_variable(tech_pvt->channel, SWITCH_LOCAL_MEDIA_IP_VARIABLE);
//const char *port = switch_channel_get_variable(tech_pvt->channel, SWITCH_LOCAL_MEDIA_PORT_VARIABLE);

4
src/switch_pcm.c
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@ -374,7 +374,7 @@ SWITCH_MODULE_LOAD_FUNCTION(core_pcm_load)
}
for (; countb > 0; countb--) {
switch_core_codec_add_implementation(pool, codec_interface,
SWITCH_CODEC_TYPE_AUDIO, 10, "L16", NULL, rate, rate, bps,
SWITCH_CODEC_TYPE_AUDIO, 70, "L16", NULL, rate, rate, bps,
mpf * countb, spf * countb, bpf * countb, ebpf * countb, 1, spf * countb,
switch_raw_init, switch_raw_encode, switch_raw_decode, switch_raw_destroy);
}
@ -391,7 +391,7 @@ SWITCH_MODULE_LOAD_FUNCTION(core_pcm_load)
for (x = 0; x < 5; x++) {
switch_core_codec_add_implementation(pool, codec_interface, SWITCH_CODEC_TYPE_AUDIO, /* enumeration defining the type of the codec */
10, /* the IANA code number */
70, /* the IANA code number */
"L16", /* the IANA code name */
NULL, /* default fmtp to send (can be overridden by the init function) */
12000, /* samples transferred per second */

12
src/switch_rtp.c
View File

@ -2427,8 +2427,13 @@ static switch_status_t read_rtp_packet(switch_rtp_t *rtp_session, switch_size_t
status = switch_socket_recvfrom(rtp_session->from_addr, rtp_session->sock_input, 0, (void *) &rtp_session->recv_msg, bytes);
ts = ntohl(rtp_session->recv_msg.header.ts);
if (*bytes ) {
if (*bytes) {
uint16_t seq = ntohs((uint16_t) rtp_session->recv_msg.header.seq);
if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BYTESWAP) && rtp_session->recv_msg.header.pt == rtp_session->rpayload) {
switch_swap_linear((int16_t *)rtp_session->recv_msg.body, (int) *bytes - rtp_header_len);
}
if (rtp_session->last_seq && rtp_session->last_seq+1 != seq) {
#ifdef DEBUG_MISSED_SEQ
@ -3708,6 +3713,11 @@ static int rtp_common_write(switch_rtp_t *rtp_session,
if (send) {
send_msg->header.seq = htons(++rtp_session->seq);
if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BYTESWAP) && send_msg->header.pt == rtp_session->payload) {
switch_swap_linear((int16_t *)send_msg->body, (int) datalen);
}
if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_SECURE_SEND)) {
int sbytes = (int) bytes;
err_status_t stat;