From 0688f61a01c11964039442dd3855c90c1cb3fd6f Mon Sep 17 00:00:00 2001 From: Sean Bright Date: Wed, 13 Sep 2017 10:38:11 -0400 Subject: chan_rtp: Use μ-law by default instead of signed linear MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 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 --- channels/chan_rtp.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) (limited to 'channels') diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c index d671706b2..2ab841480 100644 --- 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); -- cgit v1.2.3