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Automatically exported from code.google.com/p/webrtc2sip
What steps will reproduce the problem?
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Original issue reported on code.google.com by [email protected]
on 6 Jan 2013 at 8:39
What steps will reproduce the problem?
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Original issue reported on code.google.com by [email protected]
on 18 Dec 2012 at 8:04
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 6 Jan 2013 at 4:40
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
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Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:36
What steps will reproduce the problem?
I've successfully installed and run webrtc2sip on my debian6-64 and run it.
When I make a call session established and in tcdump I see packets comming to
webrtc2sip from my chrome and from legacy sip soft, but no packets go out from
webrtc2sip.
I enabled "Disable Video:" and "Enable RTCWeb Breaker" on expert page. Also in
browser js script console there is no any errors.
What version of the product are you using? On what operating system?
webrtc2sip on debian6 -64,
chrome 23.0.1271.95 m win7-64
Please provide any additional information below.
This ticket from topic
https://groups.google.com/forum/?fromgroups=#!topic/doubango/DPQbDSGsUQ0
Original issue reported on code.google.com by [email protected]
on 11 Dec 2012 at 11:39
Attachments:
What steps will reproduce the problem?
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2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
When Media Coder module is disabled only video coding is bypassed
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:34
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:30
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
https://groups.google.com/group/doubango/browse_thread/thread/809c220ee5ce67d4
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 9:54
What steps will reproduce the problem?
Try to Building the source code for Resiprocate as mentioned in the home page
1. checkout Resiprocate source code revision 9737
2. download the patch file
3. apply the as mentioned patch -p0 -i ./webrtc2sip.patch
What is the expected output? What do you see instead?
All filed patched. instead 4 files fail with the following log file
-------------------------------------------------
...
patching file resip/stack/resiprocate_10_0.vcxproj
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file resip/stack/resiprocate_10_0.vc
xproj.rej
patching file resip/stack/resiprocate_10_0.vcxproj.filters
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file resip/stack/resiprocate_10_0.vc
xproj.filters.rej
patching file resip/stack/resiprocate_8_0.vcproj
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file resip/stack/resiprocate_8_0.vcp
roj.rej
...
--------------------------------------
What version of the product are you using? On what operating system?
Cygwin on Windows XP
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 21 Sep 2012 at 3:47
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 22 Oct 2012 at 4:37
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Disconnect oldest sockets when number of count(fd) >= max(fd-size)
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:30
What steps will reproduce the problem?
1. ./repro
What is the expected output? What do you see instead?
georgesip@georgesip-HP-ProBook-4321s:/opt/resiprocate/sbin$ sudo ./repro
INFO | 20121106-151438.141 | | RESIP:DNS | 3065846272 | DnsUtil.cxx:156 |
local hostname does not contain a domain part georgesip-HP-ProBook-4321s
CRIT | 20121106-151438.143 | repro | REPRO:APP | 3065846272 |
ReproRunner.cxx:525 | In order for outbound support, the Record-Route
flow-token hack, or force-record-route to work, you MUST specify a Record-Route
URI. Launching without...
What version of the product are you using? On what operating system?
ubuntu 12.04
i just install using the guide from the website
http://code.google.com/p/webrtc2sip/
i installed in one computer and works perfectly, but in another one doesnt work
and appear the message above.
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 6 Nov 2012 at 7:19
hi all,
I checked out source code flow the path:
D:\mydoubs\doubango\branches\2.0\doubango
D:\mydoubs\iPhone\idoubs\branches\2.0
I build on xCode 4.2, iOS 5.0, LLVM GCC 4.2
But i getting error in tinySIP:
Compile tsip_header_P_Answer_State.o in .../bin/llvm-gcc-4.2 failed with exit
code 1.
Please help me.
Original issue reported on code.google.com by [email protected]
on 29 Sep 2012 at 2:44
What steps will reproduce the problem?
1.Disable video and enabled media breaker
2.Place call from the sipml5 client to pstn
3.
What is the expected output? What do you see instead?
What version of the product are you using? On what operating system?
doubango-r800, webrtc2sip-r35, freeswitch 1.3.13
Please provide any additional information below.
attached is a js, freeswitch, & webrtcgw trace
Original issue reported on code.google.com by [email protected]
on 1 Jan 2013 at 6:24
Attachments:
What steps will reproduce the problem?
1.when trying to connect SIPML5 WebRTC2Sip and Open IMS Core
2.also cant connect IMSDroid/v2.0.509 to WebRTC2SIP server
What is the expected output? What do you see instead?
sipml5 register successfully, the message "Challenging the UE",
What version of the product are you using? On what operating system?
resiprocate checkout 9737, SIPML5, Ubuntu 12.04
Please provide any additional information below.
SIP/2.0 401 Unauthorized - Challenging the UE
Via: SIP/2.0/TCP
192.168.0.104:32959;branch=z9hG4bKdaEESXJ3UpETLLW1CATOphhp6c85BvpH;rport=32959;r
eceived=192.168.0.104
Path: <sip:[email protected]:4060;lr>
Service-Route: <sip:[email protected]:6060;lr>
To: <sip:[email protected]>;tag=1312b9511db0471d35ba56752783772c-68e2
From: <sip:[email protected]>;tag=U8Ge1JdrhgxX3Nfohwfp
Call-ID: c1605d94-f01e-c15b-4a7d-dc29b5410ec9
CSeq: 35250 REGISTER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, PUBLISH,
MESSAGE, INFO
Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux))
Warning: 392 192.168.0.101:6060 "Noisy feedback tells: pid=13240
req_src_ip=192.168.0.101 req_src_port=5060 in_uri=sip:scscf.open-ims.test:6060
out_uri=sip:scscf.open-ims.test:6060 via_cnt==4"
WWW-Authenticate: Digest realm="open-ims.test",
nonce="cuvLYyH2HyVxj1wvhyNQpFHytiJeMgAA9OzN2yH3umo=", algorithm=AKAv1-MD5,
qop="auth,auth-int"
Content-Length: 0
sigcomp id=
DEBUG | 20121120-235140.549 | repro | RESIP:TRANSACTION | 3038722880 |
TuSelector.cxx:70 | Send to TU: TU: Proxy size=0
ServerTransactionTerminated daEESXJ3UpETLLW1CATOphhp6c85BvpH
DEBUG | 20121120-235140.549 | repro | REPRO:APP | 3013544768 | Proxy.cxx:154 |
Got: ServerTransactionTerminated daEESXJ3UpETLLW1CATOphhp6c85BvpH
INFO | 20121120-235140.549 | repro | REPRO:APP | 3013544768 |
RequestContext.cxx:73 | RequestContext::process(TransactionTerminated)
daEESXJ3UpETLLW1CATOphhp6c85BvpH : numtrans=2 final=1 req=SipReq: REGISTER
open-ims.test tid=daEESXJ3UpETLLW1CATOphhp6c85BvpH cseq=35250 REGISTER
[email protected] / 35250 from(wire)
DEBUG | 20121120-235140.549 | repro | RESIP:TRANSPORT | 3030330176 |
TcpBaseTransport.cxx:263 | Processing write for [ V4 192.168.0.104:32959 WS
target domain=unspecified mFlowKey=56 ]
DEBUG | 20121120-235140.549 | repro | RESIP:TRANSPORT | 3030330176 |
ConnectionManager.cxx:59 | Found fd 56
DEBUG | 20121120-235145.481 | repro | RESIP:TRANSACTION | 3038722880 |
TuSelector.cxx:70 | Send to TU: TU: Proxy size=0
ClientTransactionTerminated 26b3ed3c5484e215
DEBUG | 20121120-235145.481 | repro | REPRO:APP | 3013544768 | Proxy.cxx:154 |
Got: ClientTransactionTerminated 26b3ed3c5484e215
INFO | 20121120-235145.481 | repro | REPRO:APP | 3013544768 |
RequestContext.cxx:73 | RequestContext::process(TransactionTerminated)
26b3ed3c5484e215 : numtrans=1 final=1 req=SipReq: REGISTER open-ims.test
tid=IaiLRhL50icXx5oqB6aqyAM2JMIxeb1T cseq=35249 REGISTER
[email protected] / 35249 from(wire)
DEBUG | 20121120-235145.481 | repro | REPRO:APP | 3013544768 |
RequestContext.cxx:56 | RequestContext::~RequestContext() 0xb28004f8
DEBUG | 20121120-235145.548 | repro | RESIP:TRANSACTION | 3038722880 |
TuSelector.cxx:70 | Send to TU: TU: Proxy size=0
Original issue reported on code.google.com by [email protected]
on 20 Nov 2012 at 4:21
Currently file permissions of autogen.sh are set to read and write only. Please
add execute permissions.
Thanks,
Jeremy
Original issue reported on code.google.com by [email protected]
on 31 Dec 2012 at 9:58
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
When acting as server, the media session manager is only stared when the ACK
message is received. This looks like not working if the two clients are chrome
(I guess same result for Firefox Nightly). Is ACK message not forwarded to the
dialog?
Original issue reported on code.google.com by [email protected]
on 7 Jan 2013 at 2:53
What steps will reproduce the problem?
1. I build with latest version.
2. I config for idoubs run via TLS transport.
What is the expected output? What do you see instead?
idoubs can not connect.
What version of the product are you using? On what operating system?
idoubs 2.0; iOS 5.0
Please provide any additional information below.
I trying connect idoubs to server via TLS: port:5061, proxy:
proxy.sip.sipthor.net,
public id=sip:[email protected],
private ID = thanhhai
Realm=sip2sip.info
But connection is falis. doese anyone know, please help me.
this console log:
2012-10-05 23:23:41.351 idoubs[14178:207] idoubs2AppDelegate///:
applicationWillEnterForeground and RegistrationState=0
2012-10-05 23:23:41.360 idoubs[14178:207] NgnSipService///: register()
2012-10-05 23:23:41.360 idoubs[14178:207] NgnSipService///: Recycling the stack
2012-10-05 23:23:42.348 idoubs[14178:6007] NgnSipService///: Stack stopped
**WARN: function: "tsip_stack_stop()"
file:
"/Volumes/SOFT/Source/MyDoub/mydoubs/iPhone/idoubs/branches/2.0/ios-ngn-stack/..
/../../../../doubango/branches/2.0/doubango/tinySIP/src/tsip.c"
line: "834"
MSG: Stack already stopped
2012-10-05 23:23:42.350 idoubs[14178:207] NgnSipService///:
realm='sip2sip.info', impu='sip:[email protected]', impi='thanhhai'
2012-10-05 23:23:42.353 idoubs[14178:207] NgnSipService///: STUN=no
2012-10-05 23:23:42.354 idoubs[14178:207] NgnSipService///:
pcscf-host='proxy.sipthor.net', pcscf-port='5061', transport='TLS',
ipversion='ipv4'
interface: en0
**WARN: function: "tnet_sockfd_connectto()"
file:
"/Volumes/SOFT/Source/MyDoub/mydoubs/iPhone/idoubs/branches/2.0/ios-ngn-stack/..
/../../../../doubango/branches/2.0/doubango/tinyNET/src/tnet_utils.c"
line: "1476"
MSG:
TNET_ERROR_WOULDBLOCK/TNET_ERROR_ISCONN/TNET_ERROR_INPROGRESS/TNET_ERROR_EAGAIN
==> use tnet_sockfd_waitUntilWritable.
2012-10-05 23:23:42.871 idoubs[14178:530b] NgnSipService///: Stack started
**WARN: function: "recvData()"
file:
"/Volumes/SOFT/Source/MyDoub/mydoubs/iPhone/idoubs/branches/2.0/ios-ngn-stack/..
/../../../../doubango/branches/2.0/doubango/tinyNET/src/tnet_transport_cfsocket.
c"
line: "127"
MSG: IOCTLT returned zero for fd=29
Original issue reported on code.google.com by [email protected]
on 9 Oct 2012 at 3:53
http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-04#section-5.1
------------------
5.1. General
Each SIP message MUST be carried within a single WebSocket message,
and a WebSocket message MUST NOT contain more than one SIP message.
Because the WebSocket transport preserves message boundaries, the use
of the Content-Length header in SIP messages is optional when they
are transported using the WebSocket sub-protocol.
------------------
However webrtc2sip complains if a SIP request over WebSocket has no
Content-Lenght:
-------------
WARNING | 20121017-110236.991 | repro | RESIP:TRANSPORT | 140109811562240 |
ConnectionBase.cxx:320 | Malformed Content-Length in connection-based
transport. Not much we can do to fix this. SipMessage::Exception Missing header
Content-Length @ SipMessage.cxx:1371
-------------
Original issue reported on code.google.com by [email protected]
on 17 Oct 2012 at 4:11
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Must be configurable using the xml file
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:32
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:31
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:40
What steps will reproduce the problem?
1.we i run ../bindings/java/android/buildAll.sh srcipt
2.i use android-ndk-r4-crystax, gcc-version-4.2.1
3.
What is the expected output? What do you see instead?
What version of the product are you using? On what operating system?
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 25 Nov 2012 at 9:16
Attachments:
What steps will reproduce the problem?
1.when register user using sip2sip account using IMSDroid /v2.0.509
What is the expected output? What do you see instead?
register successfull in the webrtc server, i just got a message "Relaying
Forbidden"
What version of the product are you using? On what operating system?
webrtc2sip and ubuntu 12.04
Please provide any additional information below.
i got these output when i try to connect, that's all
SIP/2.0 403 Relaying Forbidden
Via: SIP/2.0/UDP 192.168.0.102:60339;branch=z9hG4bK446620000;rport=60339
To: <sip:[email protected]>;tag=84bcca67
From: <sip:[email protected]>;tag=1379160743
Call-ID: e241af8e-bef4-fadd-5273-1e37ee99fe9a
CSeq: 1953493554 REGISTER
Server: webrtc2sip
Content-Length: 0
sigcomp id=
Original issue reported on code.google.com by [email protected]
on 17 Nov 2012 at 9:27
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Post some short quideline http://code.google.com/p/webrtc2sip/wiki/Asterisk
like here
http://code.google.com/p/sipml5/wiki/Asterisk
configuring asterisk ( sip.conf, extensions.conf ) and (config.xml)
What version of the product are you using? On what operating system?
webrtc2sip + ASterisk11 + Chrome 23 ( windows )
Please provide any additional information below.
I am strugling some sort of problems configuring all together, Asterisk,
Webrtc2sip, To eliminate confusions of comminity.
p.s please!
Original issue reported on code.google.com by [email protected]
on 7 Jan 2013 at 3:04
What steps will reproduce the problem?
1. Use SIPML5 demo page, lightly modified to enable to initiate calls without
priori registration
2. Make a call to a SIP legacy video system with Chrome stable (M24)
3.
What is the expected output? What do you see instead?
No audio/video.
ICE connectivity checks seem to fail. Looking at network traces show that no
STUN requests are sent/received (except those sent by webrtc2sip to
stun.l.google.com). Furthermore (or consequently), webrtc2sip sends SRTP to a
candidate IP that is not routable for it (192.168.1.24).
What version of the product are you using? On what operating system?
webrtc2sip 2.2.0
SIPML5 r168
Please provide any additional information below.
webrtc2sip config.xml:
<config>
<debug-level>INFO</debug-level>
<transport>udp;*;10060</transport>
<transport>ws;*;10060</transport>
<transport>wss;*;10062</transport>
<enable-100rel>no</enable-100rel>
<enable-media-coder>yes</enable-media-coder>
<enable-videojb>no</enable-videojb>
<rtp-buffsize>65535</rtp-buffsize>
<avpf-tail-length>100;400</avpf-tail-length>
<srtp-mode>optional</srtp-mode>
<codecs>pcma;pcmu;vp8;h263</codecs>
<enable-rtp-symetric>no</enable-rtp-symetric>
<srtp-type>sdes</srtp-type>
<video-size-pref>cif</video-size-pref>
<!--nameserver>66.66.66.6</nameserver-->
<!--ssl-certificates>
self.pem;
self.pem;
*
</ssl-certificates-->
</config>
Original issue reported on code.google.com by [email protected]
on 15 Jan 2013 at 9:46
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 8 Dec 2012 at 9:11
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 14 Dec 2012 at 6:36
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 6:34
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:39
What steps will reproduce the problem?
1. Call started by caller <SipML5 client>
2. Callee answers the call <SIP Server>
3. Media packages start flowing via webrtc2sip
What is the expected output? What do you see instead?
Expected scenario;
- SIP Server received a call
- SIP Server answers the call and starts transmitting pre-recorded audio track with G.711 A-Law (pcma) codec.
- Caller listens the callee's audio message
Instead of expected scenario, during audio transmission from SIP server
received sound played like cluttered (as one of the webrtc2sip user Anton said,
i couldn't come up with better word :) ). I might phrase cluttered as a potato
robot with low on batteries from Portal game.
What version of the product are you using? On what operating system?
Product stack;
- SIP Server with G.711 A-Law (pcma) codec support
- webrtc2sip v2.0 (Running on Ubuntu 12.04 LTS)
- SipML5 live demo svn.20 (Running on Chrome)
Please provide any additional information below.
my webrtc2sip configuration;
<debug-level>ERROR</debug-level>
<transport>udp;*;10060</transport>
<transport>ws;*;10060</transport>
<transport>wss;*;10062</transport>
<enable-100rel>no</enable-100rel>
<enable-media-coder>no</enable-media-coder>
<enable-videojb>yes</enable-videojb>
<rtp-buffsize>65535</rtp-buffsize>
<avpf-tail-length>100;400</avpf-tail-length>
<srtp-mode>optional</srtp-mode>
<codecs>pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+</codecs>
my sipML5 configuration;
- Disable Video -> checked
- Enable RTCWeb Breaker -> checked
- WebSocket Server URL -> 'ws://example.com:10060'
- SIP outbound Proxy URL -> ''
Original issue reported on code.google.com by [email protected]
on 7 Jan 2013 at 11:14
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 14 Dec 2012 at 5:31
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 14 Jan 2013 at 1:34
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:40
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 23 Oct 2012 at 8:58
I trying connect idoubs to server via TLS: port:5061, proxy:
proxy.sip.sipthor.net,
public id=sip:[email protected],
private ID = thanhhai
Realm=sip2sip.info
But connection is falis. doese anyone know, please help me.
this console log:
2012-10-05 23:23:41.351 idoubs[14178:207] idoubs2AppDelegate///:
applicationWillEnterForeground and RegistrationState=0
2012-10-05 23:23:41.360 idoubs[14178:207] NgnSipService///: register()
2012-10-05 23:23:41.360 idoubs[14178:207] NgnSipService///: Recycling the stack
2012-10-05 23:23:42.348 idoubs[14178:6007] NgnSipService///: Stack stopped
**WARN: function: "tsip_stack_stop()"
file:
"/Volumes/SOFT/Source/MyDoub/mydoubs/iPhone/idoubs/branches/2.0/ios-ngn-stack/..
/../../../../doubango/branches/2.0/doubango/tinySIP/src/tsip.c"
line: "834"
MSG: Stack already stopped
2012-10-05 23:23:42.350 idoubs[14178:207] NgnSipService///:
realm='sip2sip.info', impu='sip:[email protected]', impi='thanhhai'
2012-10-05 23:23:42.353 idoubs[14178:207] NgnSipService///: STUN=no
2012-10-05 23:23:42.354 idoubs[14178:207] NgnSipService///:
pcscf-host='proxy.sipthor.net', pcscf-port='5061', transport='TLS',
ipversion='ipv4'
interface: en0
**WARN: function: "tnet_sockfd_connectto()"
file:
"/Volumes/SOFT/Source/MyDoub/mydoubs/iPhone/idoubs/branches/2.0/ios-ngn-stack/..
/../../../../doubango/branches/2.0/doubango/tinyNET/src/tnet_utils.c"
line: "1476"
MSG:
TNET_ERROR_WOULDBLOCK/TNET_ERROR_ISCONN/TNET_ERROR_INPROGRESS/TNET_ERROR_EAGAIN
==> use tnet_sockfd_waitUntilWritable.
2012-10-05 23:23:42.871 idoubs[14178:530b] NgnSipService///: Stack started
**WARN: function: "recvData()"
file:
"/Volumes/SOFT/Source/MyDoub/mydoubs/iPhone/idoubs/branches/2.0/ios-ngn-stack/..
/../../../../doubango/branches/2.0/doubango/tinyNET/src/tnet_transport_cfsocket.
c"
line: "127"
MSG: IOCTLT returned zero for fd=29
Original issue reported on code.google.com by [email protected]
on 5 Oct 2012 at 4:25
What steps will reproduce the problem?
1.when want to make a call from Chrome to IMSDroid
What is the expected output? What do you see instead?
make an audio and video communication
Bad Request - Not following indicated Service-Routes
What version of the product are you using? On what operating system?
WebRTC2SIP, Open IMS Core, Chrome 21 and IMSDroid /v2.0.509, Ubuntu 12.04
Please provide any additional information below.
SIP/2.0 400 Bad Request - Not following indicated Service-Routes
Via: SIP/2.0/TCP
192.168.0.104:46305;branch=z9hG4bK5pZCxXDCocWoAvgFVghlRdwTpSw4klIp;rport=46305;r
eceived=192.168.0.104
To: <sip:[email protected]>;tag=bbefbf7b7128ead5ec8132128b760533.ee81
From: <sip:[email protected]>;tag=89sgNfS7F8gM2KWG4DTT
Call-ID: 8fb8d6d3-5fe0-f3ce-bede-7f1b4068fc83
CSeq: 31965 INVITE
Server: Sip EXpress router (2.1.0-dev1 OpenIMSCore (i386/linux))
Warning: 392 192.168.0.104:4060 "Noisy feedback tells: pid=3072
req_src_ip=192.168.0.104 req_src_port=1060 in_uri=sip:[email protected]
out_uri=sip:[email protected] via_cnt==2"
Content-Length: 0
sigcomp id=
DEBUG | 20121125-002338.023 | repro | RESIP:TRANSPORT | 3030178624 |
TcpBaseTransport.cxx:263 | Processing write for [ V4 192.168.0.104:46305 WS
target domain=unspecified mFlowKey=30 ]
DEBUG | 20121125-002338.023 | repro | RESIP:TRANSPORT | 3030178624 |
ConnectionManager.cxx:59 | Found fd 30
DEBUG | 20121125-002338.036 | repro | RESIP:TRANSPORT | 3030178624 |
ConnectionBase.cxx:121 | In State: NewMessage
DEBUG | 20121125-002338.037 | repro | RESIP:TRANSPORT | 3030178624 |
ConnectionBase.cxx:171 | ConnectionBase::process setting source [ V4
192.168.0.104:46305 WS target domain=unspecified mFlowKey=30 ]
DEBUG | 20121125-002338.037 | repro | RESIP:TRANSPORT | 3030178624 |
Transport.cxx:382 | incoming from: [ V4 192.168.0.104:46305 WS target
domain=unspecified mFlowKey=30 ]
DEBUG | 20121125-002338.037 | repro | RESIP:TRANSPORT | 3030178624 |
ConnectionBase.cxx:429 | ##Connection: CONN_BASE: 0x9d79ce0 [ V4
192.168.0.104:46305 WS target domain=unspecified mFlowKey=30 ] received: ACK
sip:[email protected] SIP/2.0
Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bK5pZCxXDCocWoAvgFVghlRdwTpSw4klIp;rport=46305;
received=192.168.0.104
Max-Forwards: 70
To: <sip:[email protected]>;tag=bbefbf7b7128ead5ec8132128b760533.ee81
From: <sip:[email protected]>;tag=89sgNfS7F8gM2KWG4DTT
Call-ID: 8fb8d6d3-5fe0-f3ce-bede-7f1b4068fc83
CSeq: 31965 ACK
Content-Length: 0
DEBUG | 20121125-002338.037 | repro | RESIP:TRANSPORT | 3030178624 |
Connection.cxx:400 | Connection::performReads() read=368
DEBUG | 20121125-002338.037 | repro | RESIP:TRANSPORT | 3030178624 |
ConnectionBase.cxx:894 | Creating buffer for CONN_BASE: 0x9d79ce0 [ V4
192.168.0.104:46305 WS target domain=unspecified mFlowKey=30 ]
INFO | 20121125-002338.037 | repro | RESIP:TRANSPORT | 3030178624 |
TcpConnection.cxx:42 | No data ready to read
DEBUG | 20121125-002338.037 | repro | RESIP:TRANSACTION | 3038571328 |
TuSelector.cxx:70 | Send to TU: TU: Proxy size=0
ServerTransactionTerminated 5pZCxXDCocWoAvgFVghlRdwTpSw4klIp
DEBUG | 20121125-002338.037 | repro | REPRO:APP | 3013393216 | Proxy.cxx:154 |
Got: ServerTransactionTerminated 5pZCxXDCocWoAvgFVghlRdwTpSw4klIp
INFO | 20121125-002338.037 | repro | REPRO:APP | 3013393216 |
RequestContext.cxx:73 | RequestContext::process(TransactionTerminated)
5pZCxXDCocWoAvgFVghlRdwTpSw4klIp : numtrans=2 final=1 req=SipReq: INVITE
[email protected] tid=5pZCxXDCocWoAvgFVghlRdwTpSw4klIp cseq=31965 INVITE
[email protected] / 31965 from(wire)
Original issue reported on code.google.com by [email protected]
on 24 Nov 2012 at 4:35
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 9:55
What steps will reproduce the problem?
1. Build and install Doubango using procedure mentioned in the technical guide
2. Build and install WebRTC2SIP as mentioned in the guide using options
-with-doubango=/usr/local
3.
What is the expected output? What do you see instead?
- I saw error on the first step in the step when I run ./autogen.sh
- The second time the command executed without error but I could see warnings
and messages
- I executed ./configure and everything went fine
- Same with make and make install for Doubango
- Finished installation of WebRTC2SIP with parameter -with-doubango=/usr/local,
no errors again
- I do not know how to start WebRTC2SIP. I tried executing
/opt/webrtc2sip/sbin/webrtc2sip and I get the following message
error while loading shared libraries: libtinySAK.so.0: cannot open shared
object file: No such file or directory
-Another question I have is, how do I configure SIPML5 to use the WebRTC2SIp at
ip a.b.c.d?
What version of the product are you using? On what operating system?
Centos 5.5 32 Bit
Build 799
Please provide any additional information below.
I tried the same on Centos 5.5 64 and same result
Original issue reported on code.google.com by [email protected]
on 29 Dec 2012 at 9:04
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:39
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 11 Dec 2012 at 11:45
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Issues to fix:
https://bugzilla.mozilla.org/show_bug.cgi?id=828027
https://bugzilla.mozilla.org/show_bug.cgi?id=827932
Original issue reported on code.google.com by [email protected]
on 9 Jan 2013 at 10:21
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 5 Jan 2013 at 3:11
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:37
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Only when using more than one codec with dynamic code type
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:35
Feature request: make webrtc2sip look for config.xml in default paths like /etc
or /usr/local/etc/webrtc2sip (or any other path set in /etc/default/webrtc2sip
for instance) and/or add an option to set the path for the config.xml file
(like webrtc2sip -c /usr/local/etc/webrtc2sip/config.xml).
At the moment webrtc2sip only starts up when config.xml is in the same
directory as the binary because it looks for ./config.xml. This means one has
to jump through some hoops to create a startup script for webrtc2sip. Also this
isn't very UNIXy, configuration files do not belong in a /bin or /sbin
directory ;)
Merci d'avance,
Jeremy
Original issue reported on code.google.com by [email protected]
on 20 Dec 2012 at 9:18
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:32
What steps will reproduce the problem?
1. Build Doubango with libyuv
2. Build webrtc2sip
3.
What is the expected output? What do you see instead?
Expected output: webrtc2sip builds cleanly
What do you see instead:
g++ -g -O2 -o webrtc2sip webrtc2sip-mp_engine.o webrtc2sip-mp_mediaproxy.o
webrtc2sip-mp_mutex.o webrtc2sip-mp_object.o webrtc2sip-mp_peer.o
webrtc2sip-mp_proxyplugin.o webrtc2sip-mp_proxyplugin_consumer_audio.o
webrtc2sip-mp_proxyplugin_consumer_video.o webrtc2sip-mp_proxyplugin_mgr.o
webrtc2sip-mp_proxyplugin_producer_audio.o
webrtc2sip-mp_proxyplugin_producer_video.o webrtc2sip-mp_session.o
webrtc2sip-mp_session_av.o webrtc2sip-mp_wrap.o webrtc2sip-ActionConfig.o
webrtc2sip-AudioResampler.o webrtc2sip-DDebug.o webrtc2sip-MediaContent.o
webrtc2sip-MediaSessionMgr.o webrtc2sip-Msrp.o webrtc2sip-ProxyConsumer.o
webrtc2sip-ProxyPluginMgr.o webrtc2sip-ProxyProducer.o webrtc2sip-SafeObject.o
webrtc2sip-SipCallback.o webrtc2sip-SipEvent.o webrtc2sip-SipMessage.o
webrtc2sip-SipSession.o webrtc2sip-SipStack.o webrtc2sip-SipUri.o
webrtc2sip-SMSEncoder.o webrtc2sip-Xcap.o -L/usr/local/lib -L/usr/lib
-L/usr/include -ltinySAK -ltinySIP -ltinyNET -ltinyDAV -ltinyMEDIA -ltinyHTTP
-ltinyXCAP -ltinySMS -ltinyMSRP -ltinySDP -ltinyRTP -lxml2 -lpthread
/usr/local/lib/libtinyDAV.so: undefined reference to
`chromium_jpeg_CreateDecompress'
/usr/local/lib/libtinyDAV.so: undefined reference to
`chromium_jpeg_destroy_decompress'
/usr/local/lib/libtinyDAV.so: undefined reference to
`chromium_jpeg_read_raw_data'
/usr/local/lib/libtinyDAV.so: undefined reference to `chromium_jpeg_read_header'
/usr/local/lib/libtinyDAV.so: undefined reference to
`chromium_jpeg_start_decompress'
/usr/local/lib/libtinyDAV.so: undefined reference to
`chromium_jpeg_abort_decompress'
/usr/local/lib/libtinyDAV.so: undefined reference to
`chromium_jpeg_resync_to_restart'
/usr/local/lib/libtinyDAV.so: undefined reference to `chromium_jpeg_std_error'
collect2: ld returned 1 exit status
make[1]: *** [webrtc2sip] Error 1
make[1]: Leaving directory `/usr/local/src/webrtc2sip'
make: *** [all] Error 2
What version of the product are you using? On what operating system?
Doubango and webrtc2sip from svn on Ubuntu 12.04 x86_64.
Please provide any additional information below.
These references seem to be part of a patched libjpeg-turbo source. So I
downloaded libjpeg-turbo rev 856 from svn (the same one Google uses for its
WebRTC project), patched it with google.patch that you can find in the webrtc
source (trunk/third_party/libjpeg_turbo/google.patch) and build libjpeg-turbo.
Then I build Doubango and webrtc2sip again to no avail. I'm no coder so I could
be looking in the complete wrong direction.
Original issue reported on code.google.com by [email protected]
on 6 Dec 2012 at 12:51
What steps will reproduce the problem?
1.
2.
3.
What is the expected output? What do you see instead?
Please use labels and text to provide additional information.
Original issue reported on code.google.com by [email protected]
on 3 Dec 2012 at 3:38
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