Browser->Webrtc2sip via Internet (nat), Is Webrtc2sip -> SIP Server must be both Internet too ?

[Din Assegaf]

Hi

First try on local network working,

Client/Browser (192.168.1.x) -> Webrtc2sip local network
(192.168.1.x) -> SIP Server (192.168.1.x)

SIP Register working,
call audio, working two way voice, normal


Now I am trying to do via Internet
Client/Browser (192.168.4.x) via nat -> Webrtc2sip internet network
(118.x.x.x / 192.168.1.x) -> SIP Server on local network as the same
as webrtc2sip (192.168.1.x)

SIP Register working OK,

call number, stuck, no audio, no ringing, not connected at all.

is there any missing step on http://www.webrtc2sip.org/ which didnt written ?

sorry for my bad english.



last log

==session event = connecting SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
onicecandidate = closed SIPml-api.js:1
ICE GATHERING COMPLETED! SIPml-api.js:1
SEND: INVITE sip:6...@192.168.1.130 SIP/2.0
Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKBaW3cE1GbIdECfzoWf7d28pFe0oQFZ7q;rport
From: <sip:89799...@192.168.1.130>;tag=G6DcLmoBeamDQZ4sG7ki
To: <sip:6...@192.168.1.130>
Contact: "8979993336"<sip:89799...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=8979993336;ha1=70de0ecb4b8807f0037b1d38b68fe55c;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 8c328a6b-276b-0d13-3f6b-ef1ee25a5e53
CSeq: 55063 INVITE
Content-Type: application/sdp
Content-Length: 1294
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.0.0.0
Organization: Doubango Telecom

v=0
o=- 1912091158 2 IN IP4 127.0.0.1
s=Doubango Telecom - PeerConnection
t=0 0
a=group:BUNDLE audio video
m=audio 23798 RTP/SAVPF 103 104 0 8 106 105 13 126
c=IN IP4 118.96.1.75
a=rtcp:23798 IN IP4 118.96.1.75
a=candidate:3324342156 1 udp 2113937151 192.168.4.110 52277 typ host
generation 0
a=candidate:3324342156 2 udp 2113937151 192.168.4.110 52277 typ host
generation 0
a=candidate:1198352696 1 udp 1677729535 118.96.1.75 23798 typ srflx generation 0
a=candidate:1198352696 2 udp 1677729535 118.96.1.75 23798 typ srflx generation 0
a=candidate:2292618108 1 tcp 1509957375 192.168.4.110 52022 typ host
generation 0
a=candidate:2292618108 2 tcp 1509957375 192.168.4.110 52022 typ host
generation 0
a=ice-ufrag:3yYiH5Pk0e+bSgMd
a=ice-pwd:QK4ZtHySHcpU1/Y5WVQuaCAf
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:fSeX939FPfr3DfQ4YnzW3NqDz0Aweqq4jch2rBF/
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:4059709327 cname:nA3PXe5/59YPeLZn
a=ssrc:4059709327 mslabel:ocbZTwLlzgP7Xb6dxGxzuwJQOR8SnXujKpjj
a=ssrc:4059709327 label:ocbZTwLlzgP7Xb6dxGxzuwJQOR8SnXujKpjj00
SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS
df7jal23ls0d.invalid;rport;branch=z9hG4bKBaW3cE1GbIdECfzoWf7d28pFe0oQFZ7q
From: <sip:89799...@192.168.1.130>;tag=G6DcLmoBeamDQZ4sG7ki
To: <sip:6...@192.168.1.130>
Call-ID: 8c328a6b-276b-0d13-3f6b-ef1ee25a5e53
CSeq: 55063 INVITE
Content-Length: 0

SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS
df7jal23ls0d.invalid;rport;branch=z9hG4bKBaW3cE1GbIdECfzoWf7d28pFe0oQFZ7q
From: <sip:89799...@192.168.1.130>;tag=G6DcLmoBeamDQZ4sG7ki
To: <sip:6...@192.168.1.130>;tag=1355326702130
Contact: <sip:6...@192.168.1.18:10060;transport=ws;ws-src-ip=118.96.1.75;ws-src-port=23746;ws-src-proto=ws>
Call-ID: 8c328a6b-276b-0d13-3f6b-ef1ee25a5e53
CSeq: 55063 INVITE
Content-Length: 0
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
onSipEventSession func_sipml5.js:530
SIPml.Session.Event {type: "i_ao_request", description: "Trying (sent
from the Transaction Layer)", o_event: tsip_event_invite, session:
SIPml.Session.Call}
func_sipml5.js:531
==session event = i_ao_request SIPml-api.js:1
onSipEventSession func_sipml5.js:530
SIPml.Session.Event {type: "i_ao_request", description: "Ringing",
o_event: tsip_event_invite, session: SIPml.Session.Call}
func_sipml5.js:531
==session event = i_ao_request SIPml-api.js:1
<i>Remote ringing...</i>



-
Syeh Abidin

[AutoStatic]

What SIP server are you using?

Regards,

Jeremy

[Din Assegaf]

the asterisk 1.8.12

I dont know if this nat issue, how to check it


-
Syeh Abidin

[AutoStatic]

This should work. Asterisk is probably not properly configured. Are you using nat=yes together with the externip and localnet options?

[Din Assegaf]

Thanks Mr,

> Are you using nat=yes together with the externip and localnet options?

trying to use this also, no luck

full of trial and error, I have tried my webrtc2sip to sip2sip.info,
not working also.

so I think 2 think can be happen,

1. build of webrtc2sip not perfect, I suspect cause using old version
of linux (debian lenny), the other version of debian squeeze working
great.

2. firewall issue on network (on call only), which I dont know which
port to open, nat, etc.

I will try more on other server.



-
Syeh Abidin

On Thu, Dec 27, 2012 at 3:49 PM, AutoStatic <autos...@gmail.com> wrote:
> This should work. Asterisk is probably not properly configured. Are you using nat=yes together with the externip and localnet options?
>

> --
>
>

[Mamadou DIOP]

On Dec 28, 2012, at 5:21 AM, Din Assegaf <asseg...@gmail.com> wrote:

> Thanks Mr,
>
>> Are you using nat=yes together with the externip and localnet options?
> trying to use this also, no luck
>
> full of trial and error, I have tried my webrtc2sip to sip2sip.info,
> not working also.
>
> so I think 2 think can be happen,
>
> 1. build of webrtc2sip not perfect, I suspect cause using old version
> of linux (debian lenny), the other version of debian squeeze working
> great.

Please open ticket on tracker and attach server logs (debug level = INFO)

>
> 2. firewall issue on network (on call only), which I dont know which
> port to open, nat, etc.

This is easy to check: Try webrtc2sip hosted at sipml5.org + sip2sip.info.

>
> I will try more on other server.
>
>
>
> -
> Syeh Abidin
>
>
> On Thu, Dec 27, 2012 at 3:49 PM, AutoStatic <autos...@gmail.com> wrote:
>> This should work. Asterisk is probably not properly configured. Are you using nat=yes together with the externip and localnet options?
>>
>> --
>>
>>
>

> --
>
>

[Din Assegaf]

Hi Mamadou,

I am not sure yet its regarding the sipml5/webrtc2sip issue, cause in
other server build (local network) is working just fine.

> This is easy to check: Try webrtc2sip hosted at sipml5.org + sip2sip.info.

this is working.

I will try other server and will update here


-
Syeh Abidin