chan_rtp: Use μ-law by default instead of signed linear

Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.

ASTERISK-21399 #close
Reported by: Tzafrir Cohen

Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3

diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
index d671706b252a4fba9261593bb3e224352591021a..2ab84148060c8a6abd8a4c1a63ea2ef3743ea305 100644 (file)
--- a/channels/chan_rtp.c
+++ b/channels/chan_rtp.c
@@ -119,6 +119,22 @@ static int rtp_hangup(struct ast_channel *ast)
        return 0;
 }
 
+static struct ast_format *derive_format_from_cap(struct ast_format_cap *cap)
+{
+       struct ast_format *fmt = ast_format_cap_get_format(cap, 0);
+
+       if (ast_format_cap_count(cap) == 1 && fmt == ast_format_slin) {
+               /*
+                * Because we have no SDP, we must use one of the static RTP payload
+                * assignments. Signed linear @ 8kHz does not map, so if that is our
+                * only capability, we force μ-law instead.
+                */
+               fmt = ast_format_ulaw;
+       }
+
+       return fmt;
+}
+
 /*! \brief Function called when we should prepare to call the multicast destination */
 static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
 {
@@ -173,7 +189,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
 
        fmt = ast_multicast_rtp_options_get_format(mcast_options);
        if (!fmt) {
-               fmt = ast_format_cap_get_format(cap, 0);
+               fmt = derive_format_from_cap(cap);
        }
        if (!fmt) {
                ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
@@ -300,7 +316,7 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
                        goto failure;
                }
        } else {
-               fmt = ast_format_cap_get_format(cap, 0);
+               fmt = derive_format_from_cap(cap);
                if (!fmt) {
                        ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
                                args.destination);