[asterisk-dev] ast_rtp_engine api

[Michael Blake]

Hello Asterisk-dev,

I have made some changes and been tracing through things in ddd but still
can't get the UnicastRTP to process the packets that it receives.

I then add some debugging for the channel when it gets called.

Looking at the other channels, I added the rtp glue, mutexes to protect
the channels, expanded the channel_tech, passed the channel off to

/* Unicast channel driver declaration */
static struct ast_channel_tech unicast_rtp_tech = {
        .type = "UnicastRTP",
        .description = "Unicast RTP Media Channel Driver",
        .requester = unicast_rtp_request,
        .call = rtp_call,
        .hangup = rtp_hangup,
        .read = rtp_read,
        .write = rtp_write,
        .write_video = rtp_write,
        .answer = rtp_answer,
        .indicate = rtp_indicate,
        .fixup = rtp_fixup,
};

static struct ast_rtp_glue send_rtp_glue = {
   .type = "UnicastRTP",
   .get_rtp_info = my_get_rtp_peer,
   .get_vrtp_info = my_get_vrtp_peer,
   .update_peer = my_set_rtp_peer,
   .get_codec = my_get_codec,
};


So my dialplan looks like

exten => 3262,1,Answer()
exten =>
3262,n,Dial(UnicastRTP/127.0.0.1:32000/c(ulaw)e(asterisk)v(h264),,);

I just add 1 to the port for the video (i.e.32001)

so audio gets sent to 127.0.0.1:32000, and video get sent to
127.0.0.1:32001

The console shows (ignore the error - there so I don't have to crank up
the debug level).

    -- Executing [3262 at default:2] Dial("SIP/2262-00000000",
"UnicastRTP/127.0.0.1:32000/c(ulaw)e(asterisk)v(h264),,") in new stack
[Feb  6 12:51:56] ERROR[26145][C-00000000]: chan_rtp.c:408
unicast_rtp_request: UnicastRTP audio  port listening on 16452
[Feb  6 12:51:56] ERROR[26145][C-00000000]: chan_rtp.c:423
unicast_rtp_request: UnicastRTP Video port listening on 19526

Netstat shows the ports are listening

udp        0      0 127.0.0.1:16452         0.0.0.0:*
udp        0      0 127.0.0.1:19526         0.0.0.0:*

rtp set debug ip 127.0.0.1

Sent RTP packet to      127.0.0.1:32000 (type 00, seq 021479, ts 921440,
len 000160)
Sent RTP packet to      127.0.0.1:32001 (type 97, seq 011007, ts
3829534616, len 000196)

I can watch the video using gstreamer

gst-launch-1.0 -v udpsrc port=32001 caps="application/x-rtp,
clock-rate=90000, media=video, encoding-name=H264" ! rtph264depay !
h264parse ! vaapidecodebin ! xvimagesink

So now when I try and send the video to the video port I use the script

gst-launch-1.0 filesrc location=$1 ! tsparse ! tsdemux ! h264parse !
rtph264pay ! udpsink host=127.0.0.1 port=$2

where $1 is the file and $2 is the port.

No video gets sent, and netstat shows the receive queue is full.

 udp   26214656      0 127.0.0.1:19526         0.0.0.0:*

Are there any suggestions on what I should look at to enable the channel
to process the packets on the received port?

It is not clear to me what the lifecycle of a channel is and which
callbacks are necessary in order to process the received packets.

Thank you.

Michael