diff options
| author | Joshua Colp <jcolp@digium.com> | 2017-06-06 12:04:21 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2017-06-07 08:32:47 -0500 |
| commit | a34dd63eb0a961a901c7540fb02a8c8cbbf53024 (patch) | |
| tree | d3d457a1fac5c5c2b47a2ab0a9358899ac73c4cc /channels | |
| parent | 65870021dcd8033b3b33e30d1175b99fbe7f21e4 (diff) | |
pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.
PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.
This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.
ASTERISK-26996
Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_pjsip.c | 15 |
1 files changed, 14 insertions, 1 deletions
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c index 851c9135a..48778ef9d 100644 --- a/channels/chan_pjsip.c +++ b/channels/chan_pjsip.c @@ -737,11 +737,24 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast) if (!session->endpoint->asymmetric_rtp_codec && ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { - /* For maximum compatibility we ensure that the write format matches that of the received media */ + struct ast_format_cap *caps; + + /* For maximum compatibility we ensure that the formats match that of the received media */ ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n", ast_format_get_name(f->subclass.format), ast_channel_name(ast), ast_format_get_name(ast_channel_rawwriteformat(ast))); + + caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (caps) { + ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN); + ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO); + ast_format_cap_append(caps, f->subclass.format, 0); + ast_channel_nativeformats_set(ast, caps); + ao2_ref(caps, -1); + } + ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format); + ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format); if (ast_channel_is_bridged(ast)) { ast_channel_set_unbridged_nolock(ast, 1); |