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| author | Jenkins2 <jenkins2@gerrit.asterisk.org> | 2017-09-14 12:37:05 -0500 |
|---|---|---|
| committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2017-09-14 12:37:05 -0500 |
| commit | aa90b169df3f558ca9d215567891c9f9b3379f87 (patch) | |
| tree | 604a6a5616225131a14da9c9a3cce526e4b23cea | |
| parent | 35cc916e63b0c8b8181e6bbeb9b052a81af2e422 (diff) | |
| parent | 8ab2dcb4b044e0bcd21d6131d125e51c08591cff (diff) | |
Merge "chan_rtp: Use μ-law by default instead of signed linear" into 14
| -rw-r--r-- | channels/chan_rtp.c | 20 |
1 files changed, 18 insertions, 2 deletions
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); |