doubangotelecom / webrtc2sip

Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network

Source code freely provided to you by Doubango Telecom ®.
This is part of sipML5 solution and don't hesitate to test our live demo.

webrtc2sip is a smart and powerful gateway using RTCWeb and SIP to turn your browser into a phone with audio, video and SMS capabilities. The gateway allows your web browser to make and receive calls from/to any SIP-legacy network or PSTN.
As an example, you will be able to make a call from your preferred web browser to a SIP-legacy softphone (e.g. xlite) or mobile/fixed phone.
The gateway contains four modules: SIP Proxy | RTCWeb Breaker | Media Coder | Click-to-Call service..

Global architecture

SIP Proxy

The role of the SIP Proxy module is to convert the SIP transport from WebSocket protocol to UDP, TCP or TLS which are supported by all legacy networks. If your provider or hosted server supports SIP over WebSocket (e.g. Asterisk or Kamailio) then, you can bypass the module and connect the client directly to the endpoint. Bypassing the SIP Proxy is not recommended if you’re planning to use the RTCWeb Breaker or Media Coder modules as this will requires maintaining two different connections.
There are no special requirements for the end server to be able to talk to the Proxy module.

SIP Proxy architecture

RTCWeb Breaker

The RTCWeb specifications make support for ICE and DTLS/SRTP mandatory. The problem is that many SIP-legacy endpoints (e.g. PSTN network) do not support these features. It’s up to the RTCWeb Breaker to negotiate and convert the media stream to allow these two worlds to interop.
We highly recommend checking the Technical Guide to understand how to avoid security issues when using this module. For example, FreeSWITCH do not support ICE which means it requires the RTCWeb Breaker in order to be able to connect the browser to a SIP-legacy endpoint.

RTCWeb Breaker architecture

Media Coder

The RTCWeb standard defined two MTI (Mandatory To Implement) audio codecs: opus and g.711.
For now there are intense discussions about the MTI video codecs. The choice is between VP8 and H.264. VP8 is royalty-free but not widely deployed while H.264 AVC is not free but widely deployed. Google has decided to use VP8 in Chrome while Ericsson uses H.264 AVC in Bowser. Mozilla and Opera Software will probably use VP8 and Microsoft H.264 AVC. As an example, the Media Coder will allow to make video calls between Chrome and Bowser. Another example is calling a Telepresence system (e.g. Cisco) which most likely uses H.264 SVC from Chrome.

Media Coder architecture

Click-to-Call service

This is a complete SIP click-to-call solution based on the three other components. The goal is to allow any person receiving your mails, visiting your website, reading your twitts, watching your Facebook/Google+ profile to call you on your mobile phone with a single click. As an example, click here to call me on my mobile phone.
For more information: http://click2dial.org

Click-to-Call Components

Testing the gateway

Let's say the webrtc2sip gateway and SIP server are running on two different PCs with IP addresses equal to 192.168.0.1 and 192.168.0.2 respectively.

1. Open http://sipml5.org/expert.htm in your browser
2. Fill “WebSocket Server URL” field with the IP address and port where your webrtc2sip gateway is listening for incoming Websocket connections (e.g ws://192.168.0.1:10060 or wss://192.168.0.1:10062). IMPORTANT: Do not forget the url scheme (ws:// or wss://).
3. The “SIP outbound Proxy URL” is used to set the destination IP address and Port to use for all outgoing requests regardless the domain name (a.k.a realm). This is a good option for developers using a SIP domain name without valid DNS A/NAPTR/SRV records. E.g. udp://192.168.0.2:5060.
4. Check “Enable RTCWeb Breaker” if you want to call a SIP-legacy endpoint.

Security Issues

We highly recommend checking the Technical Guide to understand how to avoid security issues when using our gateway.

Technical help

Please check our issue tracker, developer group and technical guide if you have any problem.

© 2012-2015 Doubango Telecom
Inspiring the future

webrtc2sip's Issues

Codec type mismatch when bypassing is enabled

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

tinyWRAP_wrap.cxx:308: error: invalid conversion from 'void**' to 'JNIEnv**'

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:

• buildlog.txt

Allows setting srtp mode in xml conf

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

Adds support for DTLS-SRTP

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

webrtc2sip fails to build: undefined references

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

Add support for opus audio codec

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

Fail to build tinySIP

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

Connect via TLS transport

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

Content-Lenght is NOT required in SIP over WebSocket

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

error Webrtc2sip with IMSDroid /v2.0.509

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

SIP/2.0 401 Unauthorized - Challenging the UE

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

Raise event from Doubango to webrtc2sip when SIP Transport stops

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

Copy headers defined in RFC 3840/41 (caller -> calle) when the RTCWeb Breaker is eanbled

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

Raise event when network connection is down

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

Allow setting preferred video size using xml config file

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

Adds 'samples' target in the Makefile

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

Add support for Firefox native WebRTC implementation

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

Retrieve max FD_SIZE at runtime

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 3:36

Media stops after a few seconds in a call

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:

• sipml5_js_trace.txt
• fs.pcap
• webrtc2si_gw.pcap

netpacket/packet.h no such file on OS X lion

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

Must not use "a=mid:audio" without BUNDLE

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 18 Dec 2012 at 8:04

chrome <-RTCWeb Breaker-> chrome do not work if media session not started on i200

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

Fail to build the webrtc2sip code

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

Feature request: make webrtc2sip look for config.xml in default paths and/or add an option to set the path for the config.xml file

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

Add support for Bowser (Ericsson's WebRTC implementation)

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

Allow setting "rtp_symetric_enabled" using the xml config file

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

Chrome M24 and webrtc2sip : ICE issue ?

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

Detect max internal socket buffer at runtime

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

Bash-ism in autogen.sh

autogen.sh contains a bash-ism (== instead of =) so it doesn't run cleanly in 
all environments. On Ubuntu 12.04 for example the output is:
./autogen.sh: 3: [: Linux: unexpected operator

I've attached a patch.

Regards,

Jeremy

Original issue reported on code.google.com by [email protected] on 18 Jan 2013 at 8:19

Attachments:

• autogen.patch

running webrtc2sip

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

Disconnect old sockets

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

Crash on CentOS64 when browser (chrome) is refreshed without clean disconnect

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

PTHREAD_MUTEX_RECURSIVE is undefined on debian

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

Adds support for draft-hixie- thewebsocketprotocol-76

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

Add support for Firefox Nightly

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

Help needed starting webrtc2sip

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

Add support for g722 audio codec

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

Please change file permissions of autogen.sh

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

rtp packets not forwarded

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:

• webrtc2sip_issue.pcap
• putty_log.txt

Add support for audio encoding/decoding bypass

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 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

Add support for RTP timeout watcher

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

Bad Request - Not following indicated Service-Routes

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

FreeSWITCH crypto issue: "a=crypto in RTP/AVP, refer to RFC 3711"

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 8:39

Sound decode problem

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

Media SSRC in RTCP-FIR from SIP-legacy to the browser is not correct

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

webrct2sip + asterisk example wiki

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

Allow logging to a file

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

Use non-zero mask key for WebSocket messages

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

Connect TLS

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

Adds support for RTCP-NACK forwarding when video jb is disabled

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