Add option to GSM HR frames to RFC5593 representation
There are different specifications around on how a GSM-HR frame should
be encapsulated into an RTP packet. RFC5593 specifies a ToC (Table of
Contents) byte to be prepended in front of the payload data.
The two formats can be distinguished easily by their length. Then the
data can be formatted into the corresponding opposite format and vice
versa.
- Add new VTY rtp-patch options
- Add conversion function
Change-Id: Iceef19e5619f8c92dfa7c8cdecb2e9b15f0a11a1
Related: OS#3807
diff --git a/src/libosmo-mgcp/mgcp_network.c b/src/libosmo-mgcp/mgcp_network.c
index 2c86f8f..33738bc 100644
--- a/src/libosmo-mgcp/mgcp_network.c
+++ b/src/libosmo-mgcp/mgcp_network.c
@@ -43,6 +43,7 @@
#include <osmocom/mgcp/mgcp_endp.h>
#include <osmocom/mgcp/mgcp_codec.h>
#include <osmocom/mgcp/debug.h>
+#include <osmocom/codec/codec.h>
#define RTP_SEQ_MOD (1 << 16)
@@ -651,6 +652,40 @@
#endif
}
+/* There are different dialects used to format and transfer voice data. When
+ * the receiving end expects GSM-HR data to be formated after RFC 5993, this
+ * function is used to convert between RFC 5993 and TS 101318, which we normally
+ * use. */
+static void rfc5993_hr_convert(struct mgcp_endpoint *endp, char *data, int *len)
+{
+ /* NOTE: *data has an overall length of RTP_BUF_SIZE, so there is
+ * plenty of space available to store the slightly larger, converted
+ * data */
+
+ struct rtp_hdr *rtp_hdr;
+
+ OSMO_ASSERT(*len >= sizeof(struct rtp_hdr));
+ rtp_hdr = (struct rtp_hdr *)data;
+
+ if (*len == GSM_HR_BYTES + sizeof(struct rtp_hdr)) {
+ /* TS 101318 encoding => RFC 5993 encoding */
+ memmove(rtp_hdr->data + 1, rtp_hdr->data, GSM_HR_BYTES);
+ rtp_hdr->data[0] = 0x00;
+ (*len) += 1;
+
+ } else if (*len == GSM_HR_BYTES + sizeof(struct rtp_hdr) + 1) {
+ /* RFC 5993 encoding => TS 101318 encoding */
+ memmove(rtp_hdr->data, rtp_hdr->data + 1, GSM_HR_BYTES);
+ (*len) -= 1;
+ } else {
+ /* It is possible that multiple payloads occur in one RTP
+ * packet. This is not supported yet. */
+ LOGP(DRTP, LOGL_ERROR,
+ "endpoint:0x%x cannot figure out how to convert RTP packet\n",
+ ENDPOINT_NUMBER(endp));
+ }
+}
+
/* Forward data to a debug tap. This is debug function that is intended for
* debugging the voice traffic with tools like gstreamer */
static void forward_data(int fd, struct mgcp_rtp_tap *tap, const char *buf,
@@ -758,6 +793,12 @@
if (addr)
mgcp_patch_and_count(endp, rtp_state, rtp_end,
addr, buf, buflen);
+
+ if (rtp_end->rfc5993_hr_convert
+ && strcmp(conn_src->end.codec->subtype_name,
+ "GSM-HR-08") == 0)
+ rfc5993_hr_convert(endp, buf, &buflen);
+
LOGP(DRTP, LOGL_DEBUG,
"endpoint:0x%x process/send to %s %s "
"rtp_port:%u rtcp_port:%u\n",