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tmakkonen / sipcmd

sipcmd
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simple commandline example to start app properly #3

Closed nov-alex closed 10 years ago

nov-alex commented 11 years ago

Hello, tmakkonen!

I try to use your app (sipcmd) and cant understand how to start. Is it possibly to show how to connect:

  1. Direct call to 300 (192.168.1.1:5060), my num is set to 200. After 300 hook off send dtmf 1 2 3 ?

2.The same, but using EBNF ?

I try use registrar (kamailio-192.168.1.1) + 2 sip client with num 100,300. Between 100 <-> 300 conn OK.

./sipcmd -P sip -R "192.168.1.1;200;;" -x "c300"
According to wireshark no one packet has been sent ((

Thanks.

pepijndevos commented 11 years ago

Same probem. How do I get this to do anything at all?

$ ./sipcmd -u "pepijndevos@sip.voipbuster.com" -R "sip.voipbuster.com;pepijndevos;MyPassw0rd;*" -w "sip.voipbuster.com" -P sip -x "csomenumber@sip.voipbuster.com"
Starting sipcmd
in debug mode
Manager
Init
initialising SIP endpoint...
Listening for SIP signalling on 0.0.0.0:TestChanAudio
TestChanAudio
5060
SIP listener up
Using SIP registrar sip.voipbuster.com for sip:pepijndevos@sip.voipbuster.com
Created LocalEndPoint
Main
sleep 2000 ms to allow time for registration ... 
Done!
## Call ##
TestPhone::Main: calling "somenumber@sip.voipbuster.com" at Thu Nov 15 09:56:03 2012

Setting up a call to: somenumber@sip.voipbuster.com
LocalEndpoint::MakeConnection
LocalEndpointCreateConnection
LocalConnection
OnIncomingConnection: token=L8d75ccc22
connection set up to somenumber@sip.voipbuster.com
OnReleased: reason: EndedByHostOffline
TestPhone::Main: shutting down
OnReleased: reason: EndedByHostOffline
OnClearedCall
~LocalConnection
TestPhone::Main: exiting...
Exiting...
~Manager
tmakkonen commented 11 years ago

There was a bug in register implementation, try again using the description in README.

jaromrax commented 11 years ago

Hi, I search for some code like spicmd, but I miss a really basic example ... I tried the previous post' command with my values - $ ./sipcmd -u "411171@sip.odorik.cz" -R "sip.odorik.cz;411171;password;*" -P sip -x "+420608544321"

and I got

Starting sipcmd in debug mode Manager Init initialising SIP endpoint... TestChanAudio TestChanAudio Created LocalEndPoint Main TestPhone::Main: shutting down TestPhone::Main: exiting... Exiting... ~Manager

...after some time it exists. No idea what is up, how to use.

Thanks for a comment

edholland commented 11 years ago

Example of making a call

./sipcmd -P sip -u <username> -c <password> -w <server> -x "c<number>;w200;d12345"
edholland commented 11 years ago

In your specific case try ./sipcmd -P sip -u 411171 -c -w sip.odorik.cz -x "c420608544321;w200;d12345"

dark4archon commented 11 years ago

./sipcmd -P sip -l 135.60.10.21 -u "1777" -c milga -w "cc.com" -x "c197253;w5000;h" -o bluga.txt

Starting sipcmd in debug mode Manager Init 0:00.027 Opal Garbage:0x36443700 PTLib Started thread 0x7dfcf0 (31086) Opal Garbage:0x36443700 TestChanAudio TestChanAudio initialising SIP endpoint... Listening for SIP signalling on 0.0.0.0:5060 SIP listener up registered as sip:1777@cc.com Created LocalEndPoint Main Call TestPhone::Main: calling "197253" using gateway "cc.com" at Thu Apr 18 10:42:29 2013 Setting up a call to: sip:197253@cc.com LocalEndpoint::MakeConnection LocalEndpointCreateConnection LocalConnection OnIncomingConnection: token=Lb4478fb52 connection set up to sip:197253@cc.com OnReleased: reason: EndedByHostOffline Wait: waiting for 5000ms OnReleased: reason: EndedByHostOffline OnClearedCall ~LocalConnection Wait: wait done Hangup Hangup: at Thu Apr 18 10:43:07 2013 TestPhone::Main: shutting down TestPhone::Main: exiting... Exiting... ~Manager

The opal log shows the SIP register and invite messages being sent out at the same time. It does not wait for the response from the registration. I tried this successfully with PJSUA and got back a "407 Proxy Authentication Required".

0:02.107 sipcmd SIP Sending PDU (640 bytes) to: >rem=udp$204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60$ REGISTER sip:cc.com SIP/2.0 CSeq: 1 REGISTER Via: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1

0:02.211 sipcmd SIP Sending PDU (1979 bytes) to: rem=udp$204.11.192.161:5080,local=udp$135.60.10.21:5060,if=135.6$ INVITE sip:197253@cc.com SIP/2.0 CSeq: 1 INVITE v: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1

Can sipcmd do this Proxy Authentication stuff? Please help.

Thx!

edholland commented 11 years ago

Could you attach the whole opal log please. From what i remember off the top of my head there should be a flag set to force a wait after registrayion (or maybe before invites). I will try and look into this soon :)

On 18 April 2013 16:55, dark4archon notifications@github.com wrote:

./sipcmd -P sip -l 135.60.10.21 -u "1777" -c milga -w "cc.com" -x "c197253;w5000;h" -o bluga.txt

Starting sipcmd in debug mode Manager Init 0:00.027 Opal Garbage:0x36443700 PTLib Started thread 0x7dfcf0 (31086) Opal Garbage:0x36443700 TestChanAudio TestChanAudio initialising SIP endpoint... Listening for SIP signalling on 0.0.0.0:5060 SIP listener up registered as sip:1777@cc.com Created LocalEndPoint Main Call TestPhone::Main: calling "197253" using gateway "cc.com" at Thu Apr 18 10:42:29 2013 Setting up a call to: sip:197253@cc.com LocalEndpoint::MakeConnection LocalEndpointCreateConnection LocalConnection OnIncomingConnection: token=Lb4478fb52 connection set up to sip:197253@cc.com OnReleased: reason: EndedByHostOffline Wait: waiting for 5000ms OnReleased: reason: EndedByHostOffline OnClearedCall ~LocalConnection Wait: wait done Hangup Hangup: at Thu Apr 18 10:43:07 2013

TestPhone::Main: shutting down TestPhone::Main: exiting... Exiting... ~Manager

The opal log shows the SIP register and invite messages being sent out at the same time. It does not wait for the response from the registration. I tried this successfully with PJSUA and got back a "407 Proxy Authentication Required".

0:02.107 sipcmd SIP Sending PDU (640 bytes) to: >rem=udp$ 204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60$ REGISTER sip:cc.com SIP/2.0 CSeq: 1 REGISTER Via: SIP/2.0/UDP 135.60.10.21:5060 ;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1

0:02.211 sipcmd SIP Sending PDU (1979 bytes) to: rem=udp$ 204.11.192.161:5080,local=udp$135.60.10.21:5060,if=135.6$ INVITE sip:197253@cc.com SIP/2.0 CSeq: 1 INVITE v: SIP/2.0/UDP 135.60.10.21:5060 ;branch=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1

Can sipcmd do this Proxy Authentication stuff? Please help.

Thx!

— Reply to this email directly or view it on GitHubhttps://github.com/tmakkonen/sipcmd/issues/3#issuecomment-16585316 .

dark4archon commented 11 years ago

Opal log below

dark4archon commented 11 years ago

Opal Log:

0:00.027 sipcmd Version 1.0.1 by Command line VoIP testphone on Unix Linux (3.2.0-34-generic-x86_64) with PTLib (v2.10.2 (svn: 25898)) at 2013/4/18 10:42:27.853 0:00.027 sipcmd OpalMan Attached endpoint with prefix sip 0:00.027 sipcmd OpalEP Created endpoint: sip 0:00.027 sipcmd PWLib File handle high water mark set: 8 PUDPSocket 0:00.027 sipcmd IfaceMon Initial interface list: 127.0.0.1 <00-00-00-00-00-00> (lo) 135.60.10.21 <00-1E-0B-2C-62-D8> (eth0) fe80::21e:bff:fe2c:62d8 <00-1E-0B-2C-62-D8> (eth0) ::1 <00-00-00-00-00-00> (lo)

0:00.027 sipcmd PTLIB Opened NetLink socket 0:00.027 sipcmd PWLib File handle high water mark set: 13 Thread unblock pipe 0:00.027 sipcmd PTLib Created thread 0x7e2530 0:00.027 sipcmd PTLib Thread high water mark set: 3 0:00.027 sipcmd PWLib File handle high water mark set: 15 Thread unblock pipe 0:00.027 sipcmd PTLib Created thread 0x7e2650 Housekeeper 0:00.027 Network In...0x36402700 PTLib Started thread 0x7e2530 (31087) Network Interface Monitor:0x36402700 0:00.027 sipcmd PTLib No permission to set priority level 4 0:00.027 Network In...0x36402700 IfaceMon Started interface monitor thread. 0:00.028 sipcmd PTLib Thread high water mark set: 4 0:00.028 Housekeeper:0x363c1700 PTLib Started thread 0x7e2650 (31088) Housekeeper:0x363c1700 0:00.028 sipcmd OpalMan Attached endpoint with prefix sips 0:00.028 sipcmd SIP Created endpoint. 0:00.028 sipcmd OpalMan Added route "local:.=sip:" 0:00.028 sipcmd OpalMan Added route "sip:.=local:" 0:00.028 sipcmd PWLib File handle high water mark set: 16 PUDPSocket 0:00.028 sipcmd MonSock Created socket bundle for all interfaces. 0:00.028 sipcmd PWLib File handle high water mark set: 17 PUDPSocket 0:00.028 sipcmd MonSock Created bundled UDP socket 135.60.10.21:5060 0:00.028 sipcmd PWLib File handle high water mark set: 18 PUDPSocket 0:00.028 sipcmd MonSock Could not listen on fe80::21e:bff:fe2c:62d8:5060 - Invalid argument 0:00.028 sipcmd PWLib File handle high water mark set: 19 Thread unblock pipe 0:00.028 sipcmd PTLib Created thread 0x7dc660 Opal Listener 0:00.028 sipcmd PTLib Thread high water mark set: 5 0:00.028 sipcmd PTLib No permission to set priority level 4 0:00.028 Opal Liste...0x36380700 PTLib Started thread 0x7dc660 (31089) Opal Listener:0x36380700 0:00.028 sipcmd SIP Start REGISTER aor=1777 remote=cc.com local= contact= authID= realm= expire=0 restore=30 minRetry=default maxRetry=default compatibility=FullyCompliant 0:00.028 Opal Liste...0x36380700 Listen Started listening thread on udp$_:5060 0:00.029 sipcmd PWLib File handle high water mark set: 20 PUDPSocket 0:00.029 sipcmd SIP Constructed REGISTER handler for sip:1777@cc.com 0:00.029 sipcmd SIP Executing state change to Subscribing for REGISTER handler, target=sip:1777@cc.com, id=f276f43 d-aca6-e211-9bc7-001e0b2c62d8@actnew 0:00.029 sipcmd SIP Changing REGISTER handler from Unavailable to Subscribing, target=sip:1777@cc.com, id=f276f43d -aca6-e211-9bc7-001e0b2c62d8@actnew 0:00.029 sipcmd DNS SRV Lookup cc.com service _sip._udp 0:02.102 sipcmd DNS SRV Record found _sip.udp.cc.com 0:02.103 sipcmd SIP Attempting SRV record entry 0: 204.11.192.161:5080 0:02.103 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060 0:02.103 sipcmd SIP Created transport udp$204.11.192.161:5080<if=udp$:5060> 0:02.103 sipcmd OpalUDP Started connect to 204.11.192.161:5080 0:02.103 sipcmd OpalUDP Writing to interface 0 - "135.60.10.21%eth0" 0:02.104 sipcmd OpalMan Listener interfaces: associated transport=udp$135.60.10.21:5060 udp$135.60.10.21:5060,udp$[fe80::21e:bff:fe2c:62d8]:5060 0:02.105 sipcmd SIP Transaction created. 0:02.106 sipcmd DNS SRV Lookup cc.com service _sip._udp 0:02.106 sipcmd DNS SRV Record found _sip._udp.cc.com 0:02.106 sipcmd SIP Attempting SRV record entry 0: 204.11.192.160:5080 0:02.106 sipcmd SIP Transaction remote address is udp$204.11.192.160:5080 0:02.106 sipcmd SIP Set new remote address udp$204.11.192.160:5080 for transport udp$204.11.192.160:5080 0:02.107 sipcmd SIP Sending PDU (640 bytes) to: rem=udp$204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60.10.21%eth0 REGISTER sip:cc.com SIP/2.0 CSeq: 1 REGISTER Via: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1 From: sip:1777@cc.com;tag=767ff43d-aca6-e211-9bc7-001e0b2c62d8 Call-ID: f276f43d-aca6-e211-9bc7-001e0b2c62d8@actnew Organization: Command line VoIP testphone To: sip:1777@cc.com Contact: sip:1777@135.60.10.21:5060;q=1, sip:1777@[fe80::21e:bff:fe2c:62d8]:5060;q=0.500 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Expires: 3600 Content-Length: 0 Max-Forwards: 70

0:02.107 sipcmd OpalUDP Setting interface to 135.60.10.21%eth0 0:02.107 sipcmd SIP Transaction timers set: retry=10.000, completion=16.000 0:02.107 sipcmd OpalUDP Skipping incompatible interface 1 - "[fe80::21e:bff:fe2c:62d8]%eth0" 0:02.107 sipcmd OpalMan Attached endpoint with prefix local 0:02.107 sipcmd OpalEP Created endpoint: local 0:02.107 sipcmd LocalEP Created endpoint. 0:02.108 sipcmd OpalMan Set up call from local:* to sip:197253@cc.com 0:02.108 sipcmd Call Created Call[Ca3d6169a1] 0:02.108 sipcmd OpalMan Set up connection to "local:" 0:02.108 sipcmd OpalCon Created connection Call[Ca3d6169a1]-EP[Lb4478fb52] 0:02.108 sipcmd LocalCon Created connection with token "Lb4478fb52" 0:02.108 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[Ca3d6169a1]-EP[Lb4478fb52] 0:02.108 sipcmd OpalMan OnIncoming connection Call[Ca3d6169a1]-EP[Lb4478fb52] 0:02.108 sipcmd Call GetOtherPartyConnection Call[Ca3d6169a1]-EP[Lb4478fb52] 0:02.108 sipcmd OpalMan Searching for route "local:bigboss sip:197253@cc.com" 0:02.108 sipcmd OpalMan Matched regex "^local:. .$" ("local:.") 0:02.108 sipcmd OpalMan Set up connection to "sip:197253@cc.com" 0:02.109 sipcmd OpalCon Created connection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] 0:02.109 sipcmd RFC2833 Handler created 0:02.109 sipcmd RFC2833 Handler created 0:02.109 sipcmd SIP Created connection. 0:02.109 sipcmd LocalCon Outgoing call routed to sip:197253@cc.com for Call[Ca3d6169a1]-EP[Lb4478fb52] 0:02.109 sipcmd Call OnSetUp Call[Ca3d6169a1]-EP[Lb4478fb52] 0:02.109 sipcmd SIP SetUpConnection: sip:197253@cc.com 0:02.109 sipcmd DNS SRV Lookup cc.com service _sip._udp 0:02.109 sipcmd DNS SRV Record found _sip._udp.cc.com 0:02.109 sipcmd SIP Attempting SRV record entry 0: 204.11.192.170:5080 0:02.109 sipcmd SIP Connecting to sip:197253@cc.com via sip:197253@204.11.192.170:5080 0:02.109 sipcmd SIP Setting new transport for destination "sip:197253@204.11.192.170:5080" 0:02.196 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060 0:02.196 sipcmd SIP Created transport udp$204.11.192.170:5080 0:02.196 sipcmd OpalUDP Started connect to 204.11.192.170:5080 0:02.202 sipcmd Call GetMediaFormats for Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] GSM-AMR GSM-06.10 G.711-uLaw-64k G.711-ALaw-64k H.263 H.261 CELT-48K CELT-32K G.722-64k G.722.1-24k G.722.1-32k G.722.2 SILK-16 SpeexIETFWide-20.6k SpeexWB SpeexWide-20.6k G.726-16k G.726-24k G.726-32k G.726-40k LPC-10 MS-GSM MS-IMA-ADPCM SILK-8 SpeexIETFNarrow-11k SpeexIETFNarrow-15k SpeexIETFNarrow-18.2k SpeexIETFNarrow-24.6k SpeexIETFNarrow-5.95k SpeexIETFNarrow-8k SpeexNB SpeexWNarrow-8k T.38 UserInput/RFC2833 NamedSignalEvent H.261-CIF H.261-QCIF H.263P RFC4175_YCbCr-4:2:0 theora MSRP SIP-IM T.140 H.224/H323AnnexQ

0:02.202 sipcmd SIP Remote media formats set to GSM-AMR,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,H.263,H.261,CELT-48K,CELT-32K,G.72 2-64k,G.722.1-24k,G.722.1-32k,G.722.2,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,G.726-16k,G.726-24k,G.726-32k,G.726-40k,LPC-10,MS-GSM,MS-IMA-ADPCM,S ILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.3 8,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,H.263P,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ 0:02.202 sipcmd OpalUDP Writing to interface 0 - "135.60.10.21%eth0" 0:02.202 sipcmd SIP Getting local URI from registeration: sip:1777@cc.com 0:02.202 sipcmd SIP Updating dialog tag from "" to "f4d6313f-aca6-e211-9bc7-001e0b2c62d8" 0:02.203 sipcmd SIP Remote dialog address from target: sip:197253@cc.com 0:02.203 sipcmd DNS SRV Lookup cc.com service _sip._udp 0:02.203 sipcmd DNS SRV Record found _sip._udp.cc.com 0:02.203 sipcmd SIP Attempting SRV record entry 0: 204.11.192.161:5080 0:02.204 sipcmd SIP INVITE transaction id=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8 created. 0:02.204 sipcmd SIP Creating INVITE request 0:02.204 sipcmd SIP Offering all configured media: GSM-AMR,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,H.263,H.261,CELT-48K,CELT-32K,G.722-64k,G.722.1-24k,G.722.1-32k,G.722.2,SILK-16,SpeexIETFWide-20.6k,SpeexWB,Sp eexWide-20.6k,G.726-16k,G.726-24k,G.726-32k,G.726-40k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIET FNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,H.263P,RFC4175_YCb Cr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ 0:02.204 sipcmd SIP Offering media type audio in SDP 0:02.204 sipcmd Call IsMediaBypassPossible Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] session 1 0:02.204 sipcmd OpalMan IsMediaBypassPossible: session 1 0:02.204 sipcmd OpalCon IsMediaBypassPossible: default returns false 0:02.204 sipcmd RTP Cannot find media session 1 0:02.204 sipcmd RTP Cannot find RTP session 1 0:02.205 sipcmd VoIP Metrics RTCP_XR_Metrics created. 0:02.205 sipcmd RTP_UDP Session 1, created with NAT flag set to 0 0:02.205 sipcmd PWLib File handle high water mark set: 21 PUDPSocket 0:02.205 sipcmd PWLib File handle low water mark set: 20 PUDPSocket 0:02.205 sipcmd RTP_UDP Session 1 created: 135.60.10.21:5000-5001 ssrc=3349705597 0:02.205 sipcmd PWLib File handle high water mark set: 22 PUDPSocket 0:02.205 sipcmd RTP Creating new session RTP_UDP 0:02.205 sipcmd RTPEp Session 1, remembering local RTP port 5000 on connection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e 0b2c62d8] 0:02.205 sipcmd RTP Found existing media session 1 0:02.205 sipcmd MediaFormat Validation of merge for media option "BitRate" failed. 0:02.205 sipcmd SDP SDP not including SpeexIETFWide-20.6k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexWide-20.6k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including MS-GSM as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including MS-IMA-ADPCM as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-11k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-15k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-18.2k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-24.6k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-5.95k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexIETFNarrow-8k as it is not a SIP transportable format 0:02.206 sipcmd SDP SDP not including SpeexWNarrow-8k as it is not a SIP transportable format 0:02.206 sipcmd MediaFormat Merging UserInput/RFC2833 into UserInput/RFC2833 0:02.206 sipcmd RFC2833 Set tx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833 0:02.206 sipcmd RFC2833 Set rx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833 0:02.206 sipcmd MediaFormat Merging NamedSignalEvent into NamedSignalEvent 0:02.206 sipcmd RFC2833 Set tx pt=[pt=100], events="192-193" for NamedSignalEvent 0:02.206 sipcmd RFC2833 Set rx pt=[pt=100], events="192-193" for NamedSignalEvent 0:02.206 sipcmd SIP Offering media type video in SDP 0:02.206 sipcmd Call IsMediaBypassPossible Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] session 2 0:02.206 sipcmd OpalMan IsMediaBypassPossible: session 2 0:02.206 sipcmd OpalCon IsMediaBypassPossible: default returns false 0:02.206 sipcmd RTP Cannot find media session 2 0:02.206 sipcmd RTP Cannot find RTP session 2 0:02.206 sipcmd VoIP Metrics RTCP_XR_Metrics created. 0:02.206 sipcmd RTP_UDP Session 2, created with NAT flag set to 0 0:02.206 sipcmd PWLib File handle high water mark set: 23 PUDPSocket 0:02.206 sipcmd PWLib File handle low water mark set: 22 PUDPSocket 0:02.206 sipcmd RTP_UDP SetOption(8,1048576) failed, even though it said it succeeded! 0:02.206 sipcmd RTP_UDP SetOption(8,524288) failed, even though it said it succeeded! 0:02.206 sipcmd RTP_UDP SetOption(8,262144) failed, even though it said it succeeded! 0:02.206 sipcmd RTP_UDP Session 2 created: 135.60.10.21:5002-5003 ssrc=711983677 0:02.206 sipcmd PWLib File handle high water mark set: 24 PUDPSocket 0:02.207 sipcmd RTP Creating new session RTP_UDP 0:02.207 sipcmd RTPEp Session 2, remembering local RTP port 5002 on connection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e 0b2c62d8] 0:02.207 sipcmd RTP Found existing media session 2 0:02.207 sipcmd SDP SDP not including H.261-CIF as it is not a SIP transportable format 0:02.207 sipcmd SDP SDP not including H.261-QCIF as it is not a SIP transportable format 0:02.207 sipcmd SIP Transaction remote address is udp$204.11.192.161:5080 0:02.207 sipcmd SIP Set new remote address udp$204.11.192.161:5080 for transport udp$204.11.192.161:5080 0:02.207 sipcmd OpalPlugin to_customised_options: Format Name = SILK-16 Media Type = audio Payload Type = [pt=120] Encoding Name = SILK Channels (R/W) = 1 UnsignedInt Clock Rate (R/O) = 16000 UnsignedInt Complexity (R/O) = 1 UnsignedInt Frame Time (R/O) = 320 UnsignedInt Max Bit Rate (R/O) = 30000 UnsignedInt Max Frame Size (R/O) = 75 UnsignedInt Max Frames Per Packet (R/O) = 5 UnsignedInt Needs Jitter (R/O) = 1 Boolean Protocol (R/O) = SIP String Rx Frames Per Packet (R/W) = 5 UnsignedInt Tx Frames Per Packet (R/W) = 2 UnsignedInt Use DTX (R/O) = 0 FMTP name: usedtx (0) Boolean Use In-Band FEC (R/O) = 1 FMTP name: useinbandfec (1) Boolean

0:02.208 sipcmd OpalPlugin to_customised_options: Format Name = SILK-8 Media Type = audio Payload Type = [pt=119] Encoding Name = SILK Channels (R/W) = 1 UnsignedInt Clock Rate (R/O) = 8000 UnsignedInt Complexity (R/O) = 1 UnsignedInt Frame Time (R/O) = 160 UnsignedInt Max Bit Rate (R/O) = 20000 UnsignedInt Max Frame Size (R/O) = 50 UnsignedInt Max Frames Per Packet (R/O) = 5 UnsignedInt Needs Jitter (R/O) = 1 Boolean Protocol (R/O) = SIP String Rx Frames Per Packet (R/W) = 5 UnsignedInt Tx Frames Per Packet (R/W) = 2 UnsignedInt Use DTX (R/O) = 0 FMTP name: usedtx (0) Boolean Use In-Band FEC (R/O) = 1 FMTP name: useinbandfec (1) Boolean

0:02.209 sipcmd OpalPlugin to_customised_options: Format Name = H.263 Media Type = video Payload Type = H263 Encoding Name = h263 Annex F - Advanced Prediction (R/W) = 1 FMTP name: F (0) Boolean CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt CIF16 MPI (R/W) = 1 FMTP name: CIF16 (33) UnsignedInt CIF4 MPI (R/W) = 1 FMTP name: CIF4 (33) UnsignedInt Clock Rate (R/O) = 90000 UnsignedInt Content Role (R/W) = No Role Enum Content Role Mask (R/W) = 0 UnsignedInt Frame Height (R/W) = 288 UnsignedInt Frame Time (R/W) = 1500 UnsignedInt Frame Width (R/W) = 352 UnsignedInt Max Bit Rate (R/W) = 327600 UnsignedInt Max Rx Frame Height (R/W) = 1152 UnsignedInt Max Rx Frame Width (R/W) = 1408 UnsignedInt Max Tx Packet Size (R/O) = 1444 UnsignedInt MaxBR (R/W) = 0 FMTP name: maxbr (0) UnsignedInt Media Packetization (R/O) = RFC2190 String Min Rx Frame Height (R/W) = 96 UnsignedInt Min Rx Frame Width (R/W) = 128 UnsignedInt Protocol (R/O) = SIP String QCIF MPI (R/W) = 1 FMTP name: QCIF (33) UnsignedInt Rate Control Enable (R/W) = 0 Boolean Rate Controller (R/W) = String SQCIF MPI (R/W) = 1 FMTP name: SQCIF (33) UnsignedInt Target Bit Rate (R/W) = 327600 UnsignedInt Tx Key Frame Period (R/W) = 125 UnsignedInt

0:02.209 sipcmd OpalPlugin to_customised_options changed option "MaxBR" from "0" to "3276" 0:02.209 sipcmd OpalPlugin to_customised_options: Format Name = H.261 Media Type = video Payload Type = H261 Encoding Name = h261 Annex D (R/O) = 0 FMTP name: D (0) Boolean CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt Clock Rate (R/O) = 90000 UnsignedInt Content Role (R/W) = No Role Enum Content Role Mask (R/W) = 0 UnsignedInt Frame Height (R/W) = 288 UnsignedInt Frame Time (R/W) = 1500 UnsignedInt Frame Width (R/W) = 352 UnsignedInt Max Bit Rate (R/W) = 621700 UnsignedInt Max Rx Frame Height (R/O) = 288 UnsignedInt Max Rx Frame Width (R/O) = 352 UnsignedInt Max Tx Packet Size (R/O) = 1444 UnsignedInt Min Rx Frame Height (R/O) = 144 UnsignedInt Min Rx Frame Width (R/O) = 176 UnsignedInt Protocol (R/O) = SIP String QCIF MPI (R/W) = 1 FMTP name: QCIF (33) UnsignedInt Rate Control Enable (R/W) = 0 Boolean Rate Controller (R/W) = String Target Bit Rate (R/W) = 621700 UnsignedInt Tx Key Frame Period (R/W) = 125 UnsignedInt

0:02.210 sipcmd OpalPlugin to_customised_options: Format Name = H.263P Media Type = video Payload Type = [pt=94] Encoding Name = h263-1998 Annex D - Unrestricted Motion Vector (R/O) = 1 FMTP name: D (0) Boolean Annex F - Advanced Prediction (R/W) = 1 FMTP name: F (0) Boolean Annex I - Advanced INTRA Coding (R/W) = 1 FMTP name: I (0) Boolean Annex J - Deblocking Filter (R/O) = 1 FMTP name: J (0) Boolean Annex K - Slice Structure (R/O) = 0 FMTP name: K (0) UnsignedInt Annex N - Reference Picture Selection (R/O) = 0 FMTP name: N (0) Boolean Annex T - Modified Quantization (R/O) = 0 FMTP name: T (0) Boolean CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt CIF16 MPI (R/W) = 1 FMTP name: CIF16 (33) UnsignedInt CIF4 MPI (R/W) = 1 FMTP name: CIF4 (33) UnsignedInt Clock Rate (R/O) = 90000 UnsignedInt Content Role (R/W) = No Role Enum Content Role Mask (R/W) = 0 UnsignedInt Frame Height (R/W) = 288 UnsignedInt Frame Time (R/W) = 1500 UnsignedInt Frame Width (R/W) = 352 UnsignedInt Max Bit Rate (R/W) = 327600 UnsignedInt Max Rx Frame Height (R/W) = 576 UnsignedInt Max Rx Frame Width (R/W) = 704 UnsignedInt Max Tx Packet Size (R/O) = 1444 UnsignedInt Min Rx Frame Height (R/W) = 96 UnsignedInt Min Rx Frame Width (R/W) = 128 UnsignedInt Protocol (R/O) = SIP String QCIF MPI (R/W) = 1 FMTP name: QCIF (33) UnsignedInt Rate Control Enable (R/W) = 0 Boolean Rate Controller (R/W) = String SIF MPI (R/W) = 320,240,1 FMTP name: CUSTOM () String SIF4 MPI (R/W) = 640,480,1 FMTP name: CUSTOM () String SQCIF MPI (R/W) = 1 FMTP name: SQCIF (33) UnsignedInt Target Bit Rate (R/W) = 327600 UnsignedInt Tx Key Frame Period (R/W) = 125 UnsignedInt

0:02.210 sipcmd OpalPlugin to_customised_options changed option "CIF16 MPI" from "1" to "33" 0:02.210 sipcmd OpalPlugin to_customised_options: Format Name = theora Media Type = video Payload Type = [pt=125] Encoding Name = theora CAP Delivery (R/W) = in_band FMTP name: delivery-method (in_band) String CAP Height (R/W) = 576 FMTP name: height (15) UnsignedInt CAP Sampling (R/W) = YCbCr-4:2:0 FMTP name: sampling (YCbCr-4:2:0) String CAP Width (R/W) = 704 FMTP name: width (15) UnsignedInt Clock Rate (R/O) = 90000 UnsignedInt Content Role (R/W) = No Role Enum Content Role Mask (R/W) = 0 UnsignedInt Frame Height (R/W) = 288 UnsignedInt Frame Time (R/W) = 1500 UnsignedInt Frame Width (R/W) = 352 UnsignedInt Max Bit Rate (R/W) = 768000 UnsignedInt Max Rx Frame Height (R/O) = 720 UnsignedInt Max Rx Frame Width (R/O) = 1280 UnsignedInt Max Tx Packet Size (R/O) = 1444 UnsignedInt Min Rx Frame Height (R/O) = 144 UnsignedInt Min Rx Frame Width (R/O) = 176 UnsignedInt Protocol (R/O) = SIP String Rate Control Enable (R/W) = 0 Boolean Rate Controller (R/W) = String Target Bit Rate (R/W) = 768000 UnsignedInt Tx Key Frame Period (R/W) = 125 UnsignedInt

0:02.211 sipcmd SIP PDU is too large (2029 bytes) trying compact form. 0:02.211 sipcmd SIP PDU is likely too large (1979 bytes) for UDP datagram. 0:02.211 sipcmd SIP Sending PDU (1979 bytes) to: rem=udp$204.11.192.161:5080,local=udp$135.60.10.21:5060,if=135.60.10.21%eth0 INVITE sip:197253@cc.com SIP/2.0 CSeq: 1 INVITE v: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK1250403f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1 f: "bigboss" sip:1777@cc.com;tag=f4d6313f-aca6-e211-9bc7-001e0b2c62d8 i: d6db313f-aca6-e211-9bc7-001e0b2c62d8@actnew k: 100rel,replaces Organization: Command line VoIP testphone t: sip:197253@cc.com m: sip:1777@135.60.10.21 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK l: 1394 c: application/sdp Max-Forwards: 70

v=0 o=- 1366299749 1 IN IP4 135.60.10.21 s=sipcmd/1.0.1 c=IN IP4 135.60.10.21 t=0 0 m=audio 5000 RTP/AVP 112 3 0 8 114 113 9 116 117 118 120 110 124 123 122 121 111 119 109 101 100 a=sendrecv a=rtpmap:112 AMR/8000/1 a=rtpmap:3 gsm/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:114 CELT/48000/1 a=rtpmap:113 CELT/32000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:116 G7221/16000/1 a=fmtp:116 bitrate=24000 a=rtpmap:117 G7221/16000/1 a=fmtp:117 bitrate=32000 a=rtpmap:118 AMR-WB/16000/1 a=fmtp:118 octet-align=1 a=rtpmap:120 SILK/16000/1 a=rtpmap:110 Speex/16000/1 a=rtpmap:124 G726-16/8000/1 a=rtpmap:123 G726-24/8000/1 a=rtpmap:122 G726-32/8000/1 a=rtpmap:121 G726-40/8000/1 a=rtpmap:111 lpc10/8000/1 a=rtpmap:119 SILK/8000/1 a=rtpmap:109 Speex/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=maxptime:22 m=video 5002 RTP/AVP 34 31 94 97 125 b=AS:186624 b=TIAS:186624000 a=sendrecv a=rtpmap:34 h263/90000 a=fmtp:34 F=1;CIF=1;CIF16=1;CIF4=1;maxbr=3276;QCIF=1;SQCIF=1 a=rtpmap:31 h261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:94 h263-1998/90000 a=fmtp:94 D=1;F=1;I=1;J=1;CIF=1;CIF4=1;QCIF=1;CUSTOM=320,240,1;SQCIF=1 a=rtpmap:97 raw/90000 a=fmtp:97 rate=90000;height=288;width=352;colorimetry=BT601-5;depth=8;sampling=YCbCr-4:2:0 a=rtpmap:125 theora/90000 a=fmtp:125 height=576;width=704

0:02.211 sipcmd OpalUDP Setting interface to 135.60.10.21%eth0 0:02.211 sipcmd SIP Transaction timers set: retry=10.000, completion=32.000 0:02.211 sipcmd OpalUDP Skipping incompatible interface 1 - "[fe80::21e:bff:fe2c:62d8]%eth0" 0:02.212 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62 d8] 0:02.212 sipcmd OpalCon OnSetUpConnectionCall[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] 0:02.212 sipcmd OpalEP OnSetUpConnection Call[Ca3d6169a1]-EP[f4d6313f-aca6-e211-9bc7-001e0b2c62d8] 0:02.212 sipcmd OpalMan SetUpCall succeeded, call=Call[Ca3d6169a1] 0:12.108 Housekeeper:0x363c1700 SIP REGISTER transaction id=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8 timeout, making retry 1, timeout 20.000, s tate 1 0:12.108 Housekeeper:0x363c1700 SIP Set new remote address udp$204.11.192.160:5080 for transport udp$204.11.192.160:5080 0:12.108 Housekeeper:0x363c1700 SIP Sending PDU (640 bytes) to: rem=udp$204.11.192.160:5080,local=udp$135.60.10.21:5060,if=135.60.10.21%eth0 REGISTER sip:cc.com SIP/2.0 CSeq: 1 REGISTER Via: SIP/2.0/UDP 135.60.10.21:5060;branch=z9hG4bK5864313f-aca6-e211-9bc7-001e0b2c62d8;rport User-Agent: sipcmd/1.0.1 From: sip:1777@cc.com;tag=767ff43d-aca6-e211-9bc7-001e0b2c62d8 Call-ID: f276f43d-aca6-e211-9bc7-001e0b2c62d8@actnew Organization: Command line VoIP testphone To: sip:1777@cc.com Contact: sip:1777@135.60.10.21:5060;q=1, sip:1777@[fe80::21e:bff:fe2c:62d8]:5060;q=0.500 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Expires: 3600 Content-Length: 0 Max-Forwards: 70

jaromrax commented 11 years ago

Odorik: Thanks for the clear advice on comandline. Still I cannot call, but now it is more clear what happens. I get an error 401 ... SIP/2.0 401 Unauthorized. BTW, I see - it probes all interfaces, but there is only 192.168.0.136%wlan0" that makes sense.

Organization: Command line VoIP testphone To: sip:411171@sip.odorik.cz Contact: sip:411171@192.168.0.136:5060;q=1, sip:411171@192.168.137.1:5060;q=0.834, sip:411171@172.16.162.1:5060;q=0.668, sip:411171@[fe80::6a5d:43ff:fef4:8f8f]:5060;q=0.502, sip:411171@[fe80::250:56ff:fec0:1]:5060;q=0.336, sip:411171@[fe80::250:56ff:fec0:8]:5060;q=0.170 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Expires: 3600

.....It gets registered ? , but then fails

0:00.145 Opal Liste...0xb4e62b40 OpalUDP Binding to interface: 192.168.0.136:5060 0:00.145 sipcmd SIP Remote media formats set to GSM-AMR,GSM-06.10,G.711-uLaw-64k,G.711-ALaw-64k,H.263,H.261,CELT-48K,CELT-32K,G.722-64k,G.722.1-24k,G.722.1-32k,G.722.2,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,G.726-16k,G.726-24k,G.726-32k,G.726-40k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,H.263P,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ 0:00.145 sipcmd OpalUDP Writing to interface 0 - "192.168.0.136%wlan0" 0:00.145 sipcmd SIP Getting local URI from registeration: sip:411171@sip.odorik.cz 0:00.145 sipcmd SIP Updating dialog tag from "" to "8a075dbe-34a7-e211-8eac-685d43f48f8f" 0:00.145 Opal Liste...0xb4e62b40 SIP Waiting for PDU on udp$81.31.45.51:5060 0:00.145 sipcmd SIP Remote dialog address from target: sip:245373556@sip.odorik.cz 0:00.145 sipcmd DNS SRV Lookup sip.odorik.cz service _sip._udp 0:00.145 sipcmd DNS SRV Record found _sip._udp.sip.odorik.cz 0:00.145 sipcmd SIP Attempting SRV record entry 0: 81.31.45.51:5060 0:00.145 Opal Liste...0xb4e62b40 SIP PDU received: rem=udp$81.31.45.51:5060,local=udp$192.168.0.136:5060,if=192.168.0.136%wlan0 SIP/2.0 401 Unauthorized CSeq: 1 REGISTER Via: SIP/2.0/UDP 192.168.0.136:5060;branch=z9hG4bKbc095cbe-34a7-e211-8eac-685d43f48f8f;rport=5061 From: sip:411171@sip.odorik.cz;tag=c4cd58be-34a7-e211-8eac-685d43f48f8f Call-ID: e4b258be-34a7-e211-8eac-685d43f48f8f@edie To: sip:411171@sip.odorik.cz;tag=d9f55a8611ba6e9295acea97d3a19a47.d5e9 Content-Length: 0 WWW-Authenticate: Digest realm="sip.odorik.cz", nonce="UXD6klFw+WYQk+Hg9hp0c5U/Cei7JZVkQuDaqkA=", qop="auth"

0:00.145 Opal Liste...0xb4e62b40 SIP Queueing PDU "1 REGISTER <401>", transaction=z9....... well, now I dont know.......

kerrang commented 11 years ago

Hi, Every time I pull up the phone the connection is broken with: ... ... TestChannel::Write TestChannel::Write TestChannel::Write TestChannel::Write OnEstablished TestPhone::Main: shutting down OnEstablished OnEstablishedCall In call with sip:RP222250111@213.218.117.114:5060 using local:hm token=[C042e8de51] Close [ Call[C042e8de51]-EP[L6c9bcfb82] - 0xb7321870 ] TestChanAudio::CloseChannel StopAudioPlayback StopAudioRecording OnClosedMediaStream OnClosedMediaStream Close [ Call[C042e8de51]-EP[L6c9bcfb82] - 0xb7332e90 ] TestChanAudio::CloseChannel StopAudioPlayback StopAudioRecording OnClosedMediaStream OnClosedMediaStream OnReleased: reason: EndedByLocalUser ~LocalConnection OnReleased: reason: EndedByLocalUser OnClearedCall TestPhone::Main: exiting... Exiting...

What the heck?... Any suggestion? Just a little?

tmakkonen commented 10 years ago

Readme updated