contrib: Create a script that opens a SIP session before

Use the Smalltalk SIP implementation to create a call
and once the call has been established start the replay
using the commoncode. No patching of RTP occurs yet.
neels/jolly/new_handover
Holger Hans Peter Freyther 11 years ago
parent 602559fd9f
commit f42e908cea
  1. 47
      openbsc/contrib/rtp/rtp_replay_shared.st
  2. 87
      openbsc/contrib/rtp/rtp_replay_sip.st

@ -4,19 +4,58 @@ Simple UDP replay from the state files
PackageLoader fileInPackage: #Sockets.
Object subclass: SDPUtils [
"Look into using PetitParser."
SDPUtils class >> findPort: aSDP [
aSDP linesDo: [:line |
(line startsWith: 'm=audio ') ifTrue: [
| stream |
stream := line readStream
skip: 'm=audio ' size;
yourself.
^ Number readFrom: stream.
]
].
^ self error: 'Not found'.
]
SDPUtils class >> findHost: aSDP [
aSDP linesDo: [:line |
(line startsWith: 'c=IN IP4 ') ifTrue: [
| stream |
^ stream := line readStream
skip: 'c=IN IP4 ' size;
upToEnd.
]
].
^ self error: 'Not found'.
]
]
Object subclass: RTPReplay [
| filename |
| filename socket |
RTPReplay class >> on: aFile [
^ self new
initialize;
file: aFile; yourself
]
initialize [
socket := Sockets.DatagramSocket new.
]
file: aFile [
filename := aFile
]
localPort [
^ socket port
]
streamAudio: aHost port: aPort [
| file last_time last_image udp_send socket dest |
| file last_time last_image udp_send dest |
last_time := nil.
last_image := nil.
@ -24,7 +63,6 @@ Object subclass: RTPReplay [
"Send the payload"
dest := Sockets.SocketAddress byName: aHost.
socket := Sockets.DatagramSocket new.
udp_send := [:payload | | datagram |
datagram := Sockets.Datagram data: payload contents address: dest port: aPort.
socket nextPut: datagram
@ -57,7 +95,8 @@ Object subclass: RTPReplay [
"How long to wait?"
wait_image := last_image + ((time - last_time) * 1000).
[ wait_image > Time millisecondClockValue ] whileTrue: [].
[ wait_image > Time millisecondClockValue ]
whileTrue: [Processor yield].
udp_send value: data.
last_time := time.

@ -0,0 +1,87 @@
"""
Create a SIP connection and then stream...
"""
PackageLoader
fileInPackage: #OsmoSIP.
"Load for the replay code"
FileStream fileIn: 'rtp_replay_shared.st'.
Osmo.SIPCall subclass: StreamCall [
| sem stream |
createCall: aSDP [
| sdp |
stream := RTPReplay on: 'rtp_ssrc6976010.240.240.1_to_10.240.240.50.state'.
sdp := aSDP % {stream localPort}.
^ super createCall: sdp.
]
sem: aSemaphore [
sem := aSemaphore
]
sessionNew [
| host port |
Transcript nextPutAll: 'The call has started'; nl.
Transcript nextPutAll: sdp_result; nl.
host := SDPUtils findHost: sdp_result.
port := SDPUtils findPort: sdp_result.
[
stream streamAudio: host port: port.
Transcript nextPutAll: 'Streaming has finished.'; nl.
] fork.
]
sessionFailed [
sem signal
]
sessionEnd [
sem signal
]
]
Eval [
| transport agent call sem sdp_fr sdp_amr |
sdp_fr := (WriteStream on: String new)
nextPutAll: 'v=0'; cr; nl;
nextPutAll: 'o=twinkle 1739517580 1043400482 IN IP4 127.0.0.1'; cr; nl;
nextPutAll: 's=-'; cr; nl;
nextPutAll: 'c=IN IP4 127.0.0.1'; cr; nl;
nextPutAll: 't=0 0'; cr; nl;
nextPutAll: 'm=audio %1 RTP/AVP 0 101'; cr; nl;
nextPutAll: 'a=rtpmap:0 PCMU/8000'; cr; nl;
nextPutAll: 'a=rtpmap:101 telephone-event/8000'; cr; nl;
nextPutAll: 'a=fmtp:101 0-15'; cr; nl;
nextPutAll: 'a=ptime:20'; cr; nl;
contents.
sem := Semaphore new.
transport := Osmo.SIPUdpTransport
startOn: '0.0.0.0' port: 5066.
agent := Osmo.SIPUserAgent createOn: transport.
transport start.
call := (StreamCall
fromUser: 'sip:1000@sip.zecke.osmocom.org'
host: '127.0.0.1'
port: 5060
to: 'sip:123456@127.0.0.1'
on: agent)
sem: sem; yourself.
call createCall: sdp_fr.
"Wait for the stream to have ended"
sem wait.
(Delay forSeconds: 4) wait.
]
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