diff options
| author | Olle Johansson <oej@edvina.net> | 2007-02-16 13:35:44 +0000 |
|---|---|---|
| committer | Olle Johansson <oej@edvina.net> | 2007-02-16 13:35:44 +0000 |
| commit | ba32ee49d032edec3a6fc30c97cf6becee110546 (patch) | |
| tree | 896555baaa9bfecc19177b5773ace97b5092f7dc /include/asterisk/rtp.h | |
| parent | 84d1cf37fe7bba58f15945fa36d54483883a639e (diff) | |
Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk/rtp.h')
| -rw-r--r-- | include/asterisk/rtp.h | 4 |
1 files changed, 3 insertions, 1 deletions
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index 85f3b1903..5eea063db 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -75,8 +75,10 @@ struct ast_rtp_protocol { enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp); /*! Get RTP struct, or NULL if unwilling to transfer */ enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp); + /*! Get RTP struct, or NULL if unwilling to transfer */ + enum ast_rtp_get_result (* const get_trtp_info)(struct ast_channel *chan, struct ast_rtp **rtp); /*! Set RTP peer */ - int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active); + int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, struct ast_rtp *tpeer, int codecs, int nat_active); int (* const get_codec)(struct ast_channel *chan); const char * const type; AST_LIST_ENTRY(ast_rtp_protocol) list; |