drachtio-rtpengine-webrtcproxy
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drachtio
Execution can not start , refuse to connect.
drachtio config and rtpengine config | both are running in the same server and when I start to run this program getting error. ERROR, I know this is related to the port issue but I want to know how to solve it because when I change the port in rtpengine and start the service and then run this program it always gives me the same error.
drachtio-rtpengine-webrtcproxy config file
{
"drachtio": {
"host": "127.0.0.1",
"port": 9022,
"secret": "cymru"
},
"rtpengine": {
"host": "127.0.0.1",
"port": 22222,
"local-port": 2223
},
"credentials": [
{
"trunk": "my.voipprovider.com",
"auth": {
"username": "<yourusername>",
"password": "<yourpassword>"
}
}
]
}
The problem is you moved rtpengine to listen for traffic on port 2223 instead of its default port 22222:
listen-ng = 127.0.0.1:2223
You then configured your node.js application to also try to listen on that same port, which of course fails.
Leave your drachtio config file unchanged, and modify your rtpengine config to listen for ng traffic on port 2222. Again this is the default, and you caused your problems by changing this unnecessarily.
Thanks for your feedback. After moving the ng traffic on 2222, Good to see it's working. Now I'm facing another issue when I make a call which comes like
{"level":30,"time":1590591154502,"msg":"Error connecting call: Error: failed allocating endpoint from rtpengine: {\"result\":\"error\",\"error-reason\":\"Ran out of ports\"}","pid":11761,"hostname":"ip-172-31-38-128","v":1}
rtpengine is saying it ran out of ports, which is strange since it is configured to have a range of 10,000 ports. Look in /var/log/syslog for more logging from rtpengine to see if you can figure out why. Also try restarting rtpengine and running your tests again
As you told I did the same, now “ran out of the port issue is solved”, restart the rtpengine, and tried to call. It seems like the call is going but I haven’t rec. any call.
drachtio log system log drachtio-rtpengine-webrtcproxy console log1
turn drachtio server log level to debug, run the node.js app with DEBUG=drachtio:* and re-do your test. Seems like some bit of configuration between the app and the server is off
Thank you for responding! I run the node app as you said and I've gathered some logs: some of log lines from drachtio log, Can this be causing the issue?
tport_type_ws.c:541 tport_ws_deinit_secondary() 0x7ff74dcdc010 destroy wss transport 0x7ff74dcdc200
processMessageStatelessly - incoming message with call-id kjee9totqi68ttahp297 does not match an existing call leg, processed in thread 140700172697088
--Full Log List-- RTP Engine from syslog: Drachtio Server Log: App Console Log1: App Console Log2:
Your problem is with your rtpengine config. Specifically this
interface = 34.200.127.219
I can see from the drachtio logs that that address is not a local address on the server. So then, rtpengine has no ports and gives an error:
rtpengine[940]: ERR: [kjee9g4d8oigfhbt84ed]: Failed to get 2 consecutive ports on interface 34.200.127.219 for media relay (last error: Cannot assign requested address)
That address is a public address assigned to the VM, but is not explictly bound to an interface. Change your rtpengine config to this and it will probably work
interface = 172.31.38.128!34.200.127.219
Hi Dave and thanks for responding. Now outgoing calls are working but I can't hear the one-sided voice in my sip client. eg- A->B is working but B->A voice is not coming. As well as When I make a call to my SIP client, the call can't come in my SIP client.
This is this log file of drachtio. drachtio.log
you should get a network trace and study it to see if you are receiving RTP on the B leg
Hi Dave, I have checked, I think we haven't rec the RTP.
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: Closing call due to timeout
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: Final packet stats:
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: --- Tag 'cqttvb8bch', created 1:51 ago for branch '', in dialogue with 'as4b925c14'
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using **unknown codec**
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: --------- Port 127.0.0.1:30158 <> 27.62.161.65:4931 , SSRC 0, 0 p, 0 b, 0 e, 111 ts
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: --- Tag 'as4b925c14', created 1:51 ago for branch '', in dialogue with 'cqttvb8bch'
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: ------ Media #1 (audio over RTP/AVP) using **unknown codec**
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: --------- Port 127.0.0.1:30140 <> 74.120.80.132:14922, SSRC 0, 0 p, 0 b, 0 e, 111 ts
Jun 2 14:43:13 ip-172-31-38-128 rtpengine[18284]: INFO: [nln0314q7p2787d7ming]: --------- Port 127.0.0.1:30141 <> 74.120.80.132:14923 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 111 ts
---full system log--- systemlog.txt
and also in drachtio log I have noticed some lines-- These types of things are common or not? basically server automatically sends invite request eg.sip:[email protected] which is not by my side (without using any SIP client) and gives Not Found error.
drachtio:agent tokens: ["ed80abca-67f0-49bc-a5b3-e8b4f09910bc","sip","network","714","udp","37.49.230.93","65033","12:51:40.347547","999df9d3-c648-4714-a457-1a48b0d2af9d","","172.31.38.128","5060",""] +3s
drachtio:agent DrachtioAgent#_onMsg: meta: {"source":"network","address":"37.49.230.93","port":"65033","protocol":"udp","time":"12:51:40.347547","transactionId":"999df9d3-c648-4714-a457-1a48b0d2af9d","dialogId":""} +0ms
drachtio:agent Agent#handle: tracking an incoming invite with call-id 390500085-333720936-715520346, currently tracking 1 invites in progress +0ms
drachtio:srf examining INVITE, dialog id: +3s
{"level":30,"time":1591102300348,"msg":"received invite from udp/37.49.230.93:sip:[email protected] with request uri sip:[email protected]","pid":17323,"hostname":"ip-172-31-38-128","v":1}
rejecting call attempt because it is not to a registered webrtc client
drachtio:agent agent#sendResponse: {"headers":{"call-id":"390500085-333720936-715520346","cseq":"1 INVITE","from":"<sip:[email protected]>;tag=951478149","to":"<sip:[email protected]>"},"status":404,"reason":"Not Found"} +0ms
drachtio:agent sendMessage: SIP/2.0 404 Not Found
drachtio:agent Call-ID: 390500085-333720936-715520346
drachtio:agent cseq: 1 INVITE
drachtio:agent from: <sip:[email protected]>;tag=951478149
drachtio:agent to: <sip:[email protected]>
drachtio:agent Content-Length: 0
drachtio:agent
drachtio:agent +0ms
drachtio:agent opts: {"stackTxnId":"999df9d3-c648-4714-a457-1a48b0d2af9d"} +0ms
drachtio:agent Agent#sendResponse: deleted pending invite for call-id 390500085-333720936-715520346, there are now 0 pending invites +0ms
drachtio:agent wp#send 127.0.0.1:9022 - 269#f270eff5-d0a1-430b-9275-c3751eb3ef7a|sip|999df9d3-c648-4714-a457-1a48b0d2af9d|
drachtio:agent SIP/2.0 404 Not Found
drachtio:agent Call-ID: 390500085-333720936-715520346
drachtio:agent cseq: 1 INVITE
drachtio:agent from: <sip:[email protected]>;tag=951478149
--drachtio log-- drachtio.log
Is this the reason behind it? I haven't heard the one side voice or not rec. incoming call in SIP client?
Hi Dave, I have also checked in homer statistics/log and I observed "SUBSCRIBE" is also not initiate by SIP client to server. as I can see still the same error log in drachtio.
2020-06-06 04:25:42.170496 processMessageStatelessly - incoming message with call-id 1918830187-1076764588-1318441293 does not match an existing call leg, processed in thread 140620675299712
2020-06-06 04:25:42.170521 No connected clients found to handle incoming register request
2020-06-06 04:25:42.170532 processNewRequest - No providers available for REGISTER
2020-06-06 04:25:42.170653 send 288 bytes to udp/[51.178.144.219]:58549 at 04:25:42.170584:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 51.178.144.219:58549;branch=z9hG4bK1786194429;rport=58549
From: <sip:[email protected]>;tag=2095601175
To: <sip:[email protected]>;tag=FN3H2930p70gp
Call-ID: 1918830187-1076764588-1318441293
CSeq: 1 REGISTER
Content-Length: 0
--drachtio log-- drachtio.log
Is this the issue to making the call? Like I can not make a call to SIP client after registration (incoming call).