Module: sems Branch: rco/t38_relay Commit: f653884f7bc02c092ae3e044713feb8d121ead00 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sems/?a=commit;h=f653884f7bc02c092ae3e044713feb8d121ead00
Author: Raphael Coeffic <[email protected]> Committer: Raphael Coeffic <[email protected]> Date: Mon Nov 26 13:05:06 2012 +0100 adds an additional RTP relay mode: RTP_Transcoding --- core/AmB2BSession.cpp | 16 ++++++++++------ core/AmB2BSession.h | 15 ++++++++++----- 2 files changed, 20 insertions(+), 11 deletions(-) diff --git a/core/AmB2BSession.cpp b/core/AmB2BSession.cpp index bfb5d9f..8a18957 100644 --- a/core/AmB2BSession.cpp +++ b/core/AmB2BSession.cpp @@ -320,7 +320,9 @@ void AmB2BSession::onSipRequest(const AmSipRequest& req) // FIXME: update media session only in case of forwarding like when handling // replies, right? - if ((rtp_relay_mode == RTP_Relay) && media_session) { + if (((rtp_relay_mode == RTP_Relay) || + (rtp_relay_mode == RTP_Transcoding)) + && media_session) { // We have to update media session before filtering because we may want to // use the codec later filtered out for transcoding. if (parseSdp(sdp, req)) { @@ -475,7 +477,8 @@ void AmB2BSession::onSipReply(const AmSipReply& reply, AmSdp sdp; - if ((rtp_relay_mode == RTP_Relay) && + if (((rtp_relay_mode == RTP_Relay) || + (rtp_relay_mode == RTP_Transcoding)) && (reply.code >= 180 && reply.code < 300)) { // We have to update media session before filtering because we may want to @@ -734,7 +737,7 @@ int AmB2BSession::relaySip(const AmSipRequest& req) AmMimeBody r_body(req.body); const AmMimeBody* body = &r_body; - if (rtp_relay_mode == RTP_Relay && + if (((rtp_relay_mode == RTP_Relay) || (rtp_relay_mode == RTP_Transcoding)) && (req.method == SIP_METH_INVITE || req.method == SIP_METH_UPDATE || req.method == SIP_METH_ACK || req.method == SIP_METH_PRACK)) { body = req.body.hasContentType(SIP_APPLICATION_SDP); @@ -808,7 +811,7 @@ int AmB2BSession::relaySip(const AmSipRequest& orig, const AmSipReply& reply) AmMimeBody r_body(reply.body); const AmMimeBody* body = &r_body; - if (rtp_relay_mode == RTP_Relay && + if (((rtp_relay_mode == RTP_Relay) || (rtp_relay_mode == RTP_Transcoding)) && (orig.method == SIP_METH_INVITE || orig.method == SIP_METH_UPDATE || orig.method == SIP_METH_ACK || orig.method == SIP_METH_PRACK)) { body = reply.body.hasContentType(SIP_APPLICATION_SDP); @@ -877,6 +880,7 @@ void AmB2BSession::clearRtpReceiverRelay() { switch (rtp_relay_mode) { case RTP_Relay: + case RTP_Transcoding: if (media_session) { AmMediaProcessor::instance()->removeSession(media_session); @@ -1180,7 +1184,7 @@ void AmB2BCallerSession::initializeRTPRelay(AmB2BCalleeSession* callee_session) callee_session->setEnableDtmfTranscoding(enable_dtmf_transcoding); callee_session->setLowFiPLs(lowfi_payloads); - if (rtp_relay_mode == RTP_Relay) { + if ((rtp_relay_mode == RTP_Relay) || (rtp_relay_mode == RTP_Transcoding)) { setMediaSession(new AmB2BMedia(this, callee_session)); // we need to add our reference callee_session->setMediaSession(media_session); @@ -1232,7 +1236,7 @@ void AmB2BCalleeSession::onB2BEvent(B2BEvent* ev) AmMimeBody r_body(co_ev->body); const AmMimeBody* body = &co_ev->body; - if (rtp_relay_mode == RTP_Relay) { + if ((rtp_relay_mode == RTP_Relay) || (rtp_relay_mode == RTP_Transcoding)) { try { body = co_ev->body.hasContentType(SIP_APPLICATION_SDP); if (body && updateLocalBody(*body, *r_body.hasContentType(SIP_APPLICATION_SDP))) { diff --git a/core/AmB2BSession.h b/core/AmB2BSession.h index 071aabd..5f5fd36 100644 --- a/core/AmB2BSession.h +++ b/core/AmB2BSession.h @@ -122,11 +122,16 @@ class AmB2BSession: public AmSession RTP_Direct, /* audio will be realyed through us - * SDP bodies of relayed requests are filtered and connection addresses are - * replaced by us, - * additionally transcoding might be used depending on payload IDs */ - RTP_Relay - + * SDP bodies of relayed requests are filtered + * and connection addresses are replaced by us + */ + RTP_Relay, + + /* + * similar to RTP_Relay, but additionally transcoding + * might be used depending on payload IDs + */ + RTP_Transcoding }; protected: _______________________________________________ Semsdev mailing list [email protected] http://lists.iptel.org/mailman/listinfo/semsdev
