oversip send to asterisk bad registry request

[emilio defranco]

Hi,

I would like to create a simple system as:

sip web client (jssip) -> oversip -> asterisk (old version 1.4.26.2 without web socket support, I cannot shift it to more recent version because I have very complicated extensions.conf with more script called and cannot convert to new syntax).

All working on local network, no NAT.

I have mounted oversip on a pc (ip 10.0.60.151), asterisk in on another one (ip 10.0.60.130).

The client is hosted on a web server (ip 10.0.60.113), the browser i s firefox.

I tested with sipML5 but, as mentioned here https://groups.google.com/forum/#!topic/oversip/hQVZGu2j7Aw for a bug it send wrong Route header and not working.

I switch to jssip as sugested, but asterisk now reply:

2018-10-21 16:59:52] WARNING[23470]: chan_sip.c:8504 parse_register_contact: Invalid host 'c06td35e2pis.invalid'
[2018-10-21 16:59:52] WARNING[23470]: chan_sip.c:9138 register_verify: Failed to parse contact info

Is clear that oversip don't send correct data to asterisk, but I cannot seen what it are outgoing.

Every help will be appreciated.

thank in advance,

Emilio Defranco

[Iñaki Baz Castillo]

On Sun, 21 Oct 2018 at 18:20, emilio defranco <reise...@gmail.com> wrote:
> I switch to jssip as sugested, but asterisk now reply:
>
> 2018-10-21 16:59:52] WARNING[23470]: chan_sip.c:8504 parse_register_contact: Invalid host 'c06td35e2pis.invalid'
> [2018-10-21 16:59:52] WARNING[23470]: chan_sip.c:9138 register_verify: Failed to parse contact info
>
> Is clear that oversip don't send correct data to asterisk, but I cannot seen what it are outgoing.

Why is it "clear" that OverSIP does not send correct data to Asterisk?
Let's not assume wrong conclusions about the software just because
Asterisk complains. In fact, that's a bug in *Asterisk*.

Said that, please read the documentation of both JsSIP and OverSIP.
You can do tricks to avoid such a bug in *Asterisk*:

* http://oversip.net/documentation/2.0.x/api/built_in_modules/outbound_mangling/
* http://jssip.net/documentation/3.2.x/api/ua_configuration_parameters/#parameter_contact_uri


--
Iñaki Baz Castillo
<i...@aliax.net>

[emilio defranco]

Before off all, thank you for your very quickly answer.

You were right: setting up contact_uri and switch on preloaded route in jssip the client register on asterisk, without any other change on oversip (as you know, in the server.rb the setting for OverSIP::Modules::OutboundMangling are already present).

Now the problems are that asterisk fail on place a call from the web and when receive a call from other sip phone to web phone.

When I place a call from web, on the asterisk log:

[2018-10-22 16:30:48] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
[2018-10-22 16:30:48] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
[2018-10-22 16:30:48] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: audio 41410 UDP/TLS/RTP/SAVPF 109 9 0 8
[2018-10-22 16:30:49] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: video 35035 UDP/TLS/RTP/SAVPF 120 126 97
[2018-10-22 16:31:45] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
[2018-10-22 16:31:46] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
[2018-10-22 16:31:46] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: audio 40959 UDP/TLS/RTP/SAVPF 109 9 0 8
[2018-10-22 16:31:46] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: video 39278 UDP/TLS/RTP/SAVPF 120 126 97

And, when I call the web phone from another sip phone:

[2018-10-22 17:22:42] WARNING[25879] file.c: Failed to write frame
[2018-10-22 17:23:09] WARNING[30191] file.c: Failed to write frame

The oversip log (place a call):

Oct 22 16:31:34 vss151 oversip[1632]:   INFO: <SipEvents> [user] OPTIONS from sip:aste...@10.0.60.130 (UA: Asterisk PBX) to sip:2...@10.0.60.130;ov-ob=3909e93776 via UDP 10.0.60.130 : 5060
Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <OutboundMangling module> incoming Outbound flow token extracted from ;ov-ob param in RURI for SIP Request 063ebec7
Oct 22 16:31:34 vss151 oversip[1632]:   INFO: <SipEvents> [user] routing initial request to an Outbound client
Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=635
Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=550
Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=550
Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <Proxy proxy_to_users 063ebec7> received response 200
Oct 22 16:31:34 vss151 oversip[1632]:   INFO: <SipEvents> [user] incoming Outbound on_success_response: 200 'OK'
Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <SIP Request 063ebec7> forwarding response 200 "OK"
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=4094
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=4094
Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] INVITE from sip:62...@10.0.60.130 (UA: JsSIP 3.2.15) to sip:62...@10.0.60.130 via WS 10.0.60.113 : 40377
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 9094443> applying outgoing Outbound support
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <UserAssertion module> user asserted, adding P-Asserted-Identity for SIP Request 9094443
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 9094443> replying 100 "Trying"
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=242
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 9094443> trying single target: udp:10.0.60.130:5060
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <ICT 41dc5e437ede4d5db47112bd381c8574a660b6ec> sending ACK for [3456]XX response
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 9094443> received response 407
Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] on_failure_response: 407 'Proxy Authentication Required'
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 9094443> forwarding response 407 "Proxy Authentication Required"
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=456
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=421
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=421
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <IST 9094443> ACK received during completed state, now confirmed
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=4261
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=4261
Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] INVITE from sip:62...@10.0.60.130 (UA: JsSIP 3.2.15) to sip:62...@10.0.60.130 via WS 10.0.60.113 : 40377
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 3646261> applying outgoing Outbound support
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <UserAssertion module> user asserted, adding P-Asserted-Identity for SIP Request 3646261
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 3646261> replying 100 "Trying"
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=242
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 3646261> trying single target: udp:10.0.60.130:5060
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <ICT e1caee8f7ede4d5db47112bd381c8574a660b6ec> sending ACK for [3456]XX response
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 3646261> received response 488
Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] on_failure_response: 488 'Not acceptable here'
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 3646261> forwarding response 488 "Not acceptable here"
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=368
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=421
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=421
Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <IST 3646261> ACK received during completed state, now confirmed
Oct 22 16:32:06 vss151 oversip[1632]:  DEBUG: <NIST 063ebec7> timer J expires, transaction terminated
Oct 22 16:32:17 vss151 oversip[1632]:  DEBUG: <ICT 41dc5e437ede4d5db47112bd381c8574a660b6ec> timer D expires, transaction terminated
Oct 22 16:32:17 vss151 oversip[1632]:  DEBUG: <ICT e1caee8f7ede4d5db47112bd381c8574a660b6ec> timer D expires, transaction terminated

The oversip log (receving a call):

Oct 22 17:22:33 vss151 oversip[1632]:   INFO: <SipEvents> [user] INVITE from sip:62...@10.0.60.130 (UA: Asterisk PBX) to sip:62...@10.0.60.130;ov-ob=3909e93776 via UDP 10.0.60.130 : 5060
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <SIP Request 4b362503> applying outgoing Outbound support
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <OutboundMangling module> incoming Outbound flow token extracted from ;ov-ob param in RURI for SIP Request 4b362503
Oct 22 17:22:33 vss151 oversip[1632]:   INFO: <SipEvents> [user] routing initial request to an Outbound client
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <SIP Request 4b362503> replying 100 "Trying"
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=1253
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=437
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=437
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=660
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=660
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <Proxy proxy_to_users 4b362503> received response 180
Oct 22 17:22:33 vss151 oversip[1632]:  DEBUG: <SIP Request 4b362503> forwarding response 180 "Ringing"
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=465
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=465
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <ICT d92215ea2f7f70afa11e4b2883a2de47f5c188fb> sending ACK for [3456]XX response
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=365
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <Proxy proxy_to_users 4b362503> received response 488
Oct 22 17:22:37 vss151 oversip[1632]:   INFO: <SipEvents> [user] incoming Outbound on_failure_response: 488 'Not Acceptable Here'
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <SIP Request 4b362503> forwarding response 488 "Not Acceptable Here"
Oct 22 17:22:37 vss151 oversip[1632]:  DEBUG: <IST 4b362503> ACK received during completed state, now confirmed
Oct 22 17:22:41 vss151 oversip[1632]:  DEBUG: <NIST 56746150> timer J expires, transaction terminated
Oct 22 17:22:42 vss151 oversip[1632]:  DEBUG: <IST 4b362503> timer I expires, transaction terminated

I know it is not a matter strictly linked to oversip, but your suggestion is very welcome.

Thank in advance,

Emilio

[Iñaki Baz Castillo]

Asterisk 1.4.X does not support WebRTC media. OverSIP just adds
support for SIP over WebSocket, but it does not acts as a "plain RTP
<-> WebRTC RTP" gateway.

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[emilio defranco]

Ok, I know that asterisk 1.4.x don't support WebRTC, for this I implemented a system with oversip, but I understand that it is not enough.

I found janus ... maybe this component, with oversip can resolve my problem.

The idea is that oversip manage the SIP signaling and janus the media ... I am looking forward on google for info about similar implementation ...

I am very gratefull to you for the time that you dedicated to me and to my problem.

Thank You very much,

emilio

On Monday, October 22, 2018 at 6:43:19 PM UTC+2, Iñaki Baz Castillo wrote:

Asterisk 1.4.X does not support WebRTC media. OverSIP just adds
support for SIP over WebSocket, but it does not acts as a "plain RTP
<-> WebRTC RTP" gateway.
On Mon, 22 Oct 2018 at 18:39, emilio defranco <reise...@gmail.com> wrote:
>
> Before off all, thank you for your very quickly answer.
>
> You were right: setting up contact_uri and switch on preloaded route in jssip the client register on asterisk, without any other change on oversip (as you know, in the server.rb the setting for OverSIP::Modules::OutboundMangling are already present).
>
> Now the problems are that asterisk fail on place a call from the web and when receive a call from other sip phone to web phone.
>
> When I place a call from web, on the asterisk log:
>
> [2018-10-22 16:30:48] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
> [2018-10-22 16:30:48] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
> [2018-10-22 16:30:48] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: audio 41410 UDP/TLS/RTP/SAVPF 109 9 0 8
> [2018-10-22 16:30:49] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: video 35035 UDP/TLS/RTP/SAVPF 120 126 97
> [2018-10-22 16:31:45] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
> [2018-10-22 16:31:46] WARNING[23470] chan_sip.c: Too many SDP lines. Ignoring.
> [2018-10-22 16:31:46] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: audio 40959 UDP/TLS/RTP/SAVPF 109 9 0 8
> [2018-10-22 16:31:46] WARNING[23470] chan_sip.c: Unsupported SDP media type in offer: video 39278 UDP/TLS/RTP/SAVPF 120 126 97
>
> And, when I call the web phone from another sip phone:
>
> [2018-10-22 17:22:42] WARNING[25879] file.c: Failed to write frame
> [2018-10-22 17:23:09] WARNING[30191] file.c: Failed to write frame
>
> The oversip log (place a call):
>

> Oct 22 16:31:34 vss151 oversip[1632]:   INFO: <SipEvents> [user] OPTIONS from sip:as...@10.0.60.130 (UA: Asterisk PBX) to sip...@10.0.60.130;ov-ob=3909e93776 via UDP 10.0.60.130 : 5060

> Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <OutboundMangling module> incoming Outbound flow token extracted from ;ov-ob param in RURI for SIP Request 063ebec7
> Oct 22 16:31:34 vss151 oversip[1632]:   INFO: <SipEvents> [user] routing initial request to an Outbound client
> Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=635
> Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=550
> Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=550
> Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <Proxy proxy_to_users 063ebec7> received response 200
> Oct 22 16:31:34 vss151 oversip[1632]:   INFO: <SipEvents> [user] incoming Outbound on_success_response: 200 'OK'
> Oct 22 16:31:34 vss151 oversip[1632]:  DEBUG: <SIP Request 063ebec7> forwarding response 200 "OK"
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=4094
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=4094

> Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] INVITE from sip:...@10.0.60.130 (UA: JsSIP 3.2.15) to sip:...@10.0.60.130 via WS 10.0.60.113 : 40377

> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 9094443> applying outgoing Outbound support
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <UserAssertion module> user asserted, adding P-Asserted-Identity for SIP Request 9094443
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 9094443> replying 100 "Trying"
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=242
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 9094443> trying single target: udp:10.0.60.130:5060
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <ICT 41dc5e437ede4d5db47112bd381c8574a660b6ec> sending ACK for [3456]XX response
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 9094443> received response 407
> Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] on_failure_response: 407 'Proxy Authentication Required'
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 9094443> forwarding response 407 "Proxy Authentication Required"
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=456
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=421
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=421
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <IST 9094443> ACK received during completed state, now confirmed
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=4261
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=4261

> Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] INVITE from sip:...@10.0.60.130 (UA: JsSIP 3.2.15) to sip:...@10.0.60.130 via WS 10.0.60.113 : 40377

> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 3646261> applying outgoing Outbound support
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <UserAssertion module> user asserted, adding P-Asserted-Identity for SIP Request 3646261
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 3646261> replying 100 "Trying"
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=242
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 3646261> trying single target: udp:10.0.60.130:5060
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <ICT e1caee8f7ede4d5db47112bd381c8574a660b6ec> sending ACK for [3456]XX response
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <Proxy proxy_out 3646261> received response 488
> Oct 22 16:31:45 vss151 oversip[1632]:   INFO: <SipEvents> [user] on_failure_response: 488 'Not acceptable here'
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <SIP Request 3646261> forwarding response 488 "Not acceptable here"
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> sending text frame: payload_length=368
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsFraming> received text frame: FIN=true, RSV1-3=false/false/false, payload_length=421
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <WsSipApp> received WS message: type=text, length=421
> Oct 22 16:31:45 vss151 oversip[1632]:  DEBUG: <IST 3646261> ACK received during completed state, now confirmed
> Oct 22 16:32:06 vss151 oversip[1632]:  DEBUG: <NIST 063ebec7> timer J expires, transaction terminated
> Oct 22 16:32:17 vss151 oversip[1632]:  DEBUG: <ICT 41dc5e437ede4d5db47112bd381c8574a660b6ec> timer D expires, transaction terminated
> Oct 22 16:32:17 vss151 oversip[1632]:  DEBUG: <ICT e1caee8f7ede4d5db47112bd381c8574a660b6ec> timer D expires, transaction terminated
>
> The oversip log (receving a call):
>

> Oct 22 17:22:33 vss151 oversip[1632]:   INFO: <SipEvents> [user] INVITE from sip:...@10.0.60.130 (UA: Asterisk PBX) to sip:...@10.0.60.130;ov-ob=3909e93776 via UDP 10.0.60.130 : 5060

[Iñaki Baz Castillo]

If you just need a pure SIP+SDP solution you may try with a Kamailio
or OpenSIPS with RTPEngine (which does support plain RTP/SDP <->
WebRTC RTP/SDP gateway.