Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Juha Heinanen]

Shahid Hussain writes:

> Following are  the REGISTER and response messages. Is it possible to
> confirm the JSSIP client has full implementation of SIP outbound?

Looks like it if you define two sockets.

-- Juha

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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Juha Heinanen]

Due to other tests, I had missed baresip account's ;outbound paramater.
Once I added it, also reg-id was added.

-- Juha

WSS 192.168.43.160:50442 -> 192.168.43.160:5061
REGISTER sip:test.tutpro.com SIP/2.0
Via: SIP/2.0/WSS 127.0.0.1:9;branch=z9hG4bK5a4ad01f9164d358;rport
Contact: 
;expires=30;+sip.instance="";q=0.5;reg-id=1
Max-Forwards: 70
Route: 
To: 
From: "Test Account" ;tag=904f27750c59fc5b
Call-ID: 69f4eea174d327eb
CSeq: 29225 REGISTER
User-Agent: baresip v1.1.0 (x86_64/linux)
Supported: gruu, outbound, path
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER
Content-Length: 0

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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Juha Heinanen]

Olle E. Johansson writes:

> Full support for SIP outbound (using REG-id when registering etc).
> Last time I looked we did not have all nuts and bolts for it, but
> let’s give it a try.

Yes, reg-id is missing from contact.  It would be good to add so that
sip proxy can detect if registration is a new one or refreshes the old
one.

But if a connection oriented transport is used and connection breaks,
kamailio should unregister the contact.  If it does, missing of reg-id
does cause a problem.

For redundancy support, instance-id is needed and it is there.

-- Juha

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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Juha Heinanen]

Olle E. Johansson writes:

> > Have you checked baresip?
> 
> I don’t recall baresip having a full SIP outbound implementation.

baresip is able to register with two outbound proxies and supports gruu
(below). What else is needed?

-- Juha

#
TLS 192.168.43.160:49556 -> 192.168.43.160:5061
REGISTER sip:test.tutpro.com SIP/2.0
Via: SIP/2.0/TLS 127.0.0.1:5047;branch=z9hG4bK305c4adf9d548df2;rport
Contact: 
;expires=600;+sip.instance="";q=0.5
Max-Forwards: 70
To: 
From: "Test Account" ;tag=f2465ebef6a435a2
Call-ID: ea13feabfb0effd8
CSeq: 1420 REGISTER
User-Agent: baresip v1.1.0 (x86_64/linux)
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER
Content-Length: 0
#
TLS 192.168.43.160:5061 -> 192.168.43.160:49556
SIP/2.0 200 OK
Via: SIP/2.0/TLS 
127.0.0.1:5047;branch=z9hG4bK305c4adf9d548df2;rport=49556;received=192.168.43.160
To: ;tag=75fad08a1e94bbeeabf349d4bd14beda.b29c1116
From: "Test Account" ;tag=f2465ebef6a435a2
Call-ID: ea13feabfb0effd8
CSeq: 1420 REGISTER
Contact: 
;q=0.5;expires=600;received="sip:192.168.43.160:49556;transport=tls";+sip.instance=""
Supported: outbound
Server: OpenSIPg SIP Proxy (5.6.0-0b02 (x86_64/linux))
Content-Length: 
#
TLS 192.168.43.160:49556 -> 192.168.43.160:5061
INVITE sip:f...@test.tutpro.com SIP/2.0
Via: SIP/2.0/TLS 127.0.0.1:5047;branch=z9hG4bK5b8f364c3b95c0b1;rport
Contact: 
Max-Forwards: 70
To: 
From: "Test Account" ;tag=ffb1ab9552ce5a91
Call-ID: 1a70eaa0d9e358bb
CSeq: 2779 INVITE
User-Agent: baresip v1.1.0 (x86_64/linux)
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER
Supported: gruu
Content-Type: application/sdp
Content-Length: 995

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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Juha Heinanen]

Olle E. Johansson writes:

> Full implementation of SIP outbound is the only solution close to
> solving this problem in the IETF standards. 
> However, I have seen no single SIP client that have implemented this,
> even though Kamailio supports 
> it on the server side. The idea is that you always have two TCP
> connections to two active servers.

Have you checked baresip?

-- Juha

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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Shahid Hussain]

I am using a JSSIP client and they claim to be implemented RFC-5626.
Following are  the REGISTER and response messages. Is it possible to
confirm the JSSIP client has full implementation of SIP outbound?
If it supports fully then I can debug outbound and gruu functionality at
Kamailio(I have it enabled but it didn't work, have to see if I need to
enable any more settings).

REGISTER sip:sip.engageio.com SIP/2.0
Via: SIP/2.0/WSS 5qfe1k7uvuaf.invalid;branch=z9hG4bK563864
Max-Forwards: 69
To: 
From: "123456543" ;tag=em9kvbdqg2
Call-ID: 071s1uno9bk7ie1hf0p3dn
CSeq: 1 REGISTER
Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.7.5
Content-Length: 0


SIP/2.0 200 OK
Via: SIP/2.0/WSS
5qfe1k7uvuaf.invalid;branch=z9hG4bK563864;received=172.27.6.98
To: ;tag=211517f31d5fe3f765e28c150e460fc3.0b7e
From: "123456543" ;tag=em9kvbdqg2
Call-ID: 071s1uno9bk7ie1hf0p3dn
CSeq: 1 REGISTER
Contact: ;expires=600;pub-gruu="sip:123456543_28i...@sip.engageio.com
;gr=urn:uuid:b7d40d83-b1f4-4419-bced-4c464507dea6";temp-gruu="
sip:uloc-60dc6c8a-7464-1-2dfe...@sip.engageio.com
;gr";+sip.instance="";reg-id=1
Require: outbound
Supported: outbound
Server: kamailio (5.3.2 (x86_64/linux))
Content-Length: 0



On Wed, Jun 30, 2021 at 4:11 PM Juha Heinanen  wrote:

> Due to other tests, I had missed baresip account's ;outbound paramater.
> Once I added it, also reg-id was added.
>
> -- Juha
>
> WSS 192.168.43.160:50442 -> 192.168.43.160:5061
> REGISTER sip:test.tutpro.com SIP/2.0
> Via: SIP/2.0/WSS 127.0.0.1:9;branch=z9hG4bK5a4ad01f9164d358;rport
> Contact:  ;transport=wss>;expires=30;+sip.instance="";q=0.5;reg-id=1
> Max-Forwards: 70
> Route: 
> To: 
> From: "Test Account" ;tag=904f27750c59fc5b
> Call-ID: 69f4eea174d327eb
> CSeq: 29225 REGISTER
> User-Agent: baresip v1.1.0 (x86_64/linux)
> Supported: gruu, outbound, path
> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER
> Content-Length: 0
>
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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Daniel-Constantin Mierla]

On 30.06.21 09:28, Juha Heinanen wrote:
> Shahid Hussain writes:
>
>> Would like to know what is the recommended solution for this problem using
>> alias or is it a limitation of using alias?
> Maybe a limitation.  Try with SIP User Agents that support gruu and thus
> identify themselves using sip.instance.

Relying on GRUU would be ok, but most of the UAs I had to deal with lack
its implementation.

What I have seen as an alternative (e.g., linphone, iirc) is sending
re-INVITE with the new contact address and then a new alias will be
generated as well. It is not enough for the UA to send a new REGISTER,
that is for receiving new calls. In making calls is not restricted to
registered devices, therefore when an UA changes is local
states/coordinates, it has to send a re-INVITE. Contact can be changed
within dialog (also, just for sake of completion, dialog module should
save new Contact values), only Record-Route/Route headers are fixed for
active calls (as far as I remember right now).

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda


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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Olle E. Johansson]

> On 30 Jun 2021, at 10:14, Juha Heinanen  wrote:
> 
> Olle E. Johansson writes:
> 
>>> Have you checked baresip?
>> 
>> I don’t recall baresip having a full SIP outbound implementation.
> 
> baresip is able to register with two outbound proxies and supports gruu
> (below). What else is needed?

Full support for SIP outbound (using REG-id when registering etc).
Last time I looked we did not have all nuts and bolts for it, but let’s give it 
a try.

/O
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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Olle E. Johansson]

> On 30 Jun 2021, at 09:49, Juha Heinanen  wrote:
> 
> Olle E. Johansson writes:
> 
>> Full implementation of SIP outbound is the only solution close to
>> solving this problem in the IETF standards. 
>> However, I have seen no single SIP client that have implemented this,
>> even though Kamailio supports 
>> it on the server side. The idea is that you always have two TCP
>> connections to two active servers.
> 
> Have you checked baresip?

I don’t recall baresip having a full SIP outbound implementation.

SIP outbound seems to be an elegant solution to a non-problem. Maybe it’s 
coming back on the radar now, several years later and we need to start working 
on it as more and more SIP moves to TCP/TLS or WebSocket and network operators 
implement CGnat and other middleboxes that do strange stuff to open flows.

Without SIP outbound a SIP UA using TLS needs to have a server cert and accept 
incoming TLS connections, which of course does not work over NAT, so if we want 
to be RFC compliant, we should at least copy the RFC for SIP over WebSocket 
“half-outbound” requirement to SIP/TLS and SIP/TCP.

I have started discussions about this “half-outbound” concept a few times, but 
haven’t gotten much support in the SIPcore IETF group. Meanwhile, Kamailio 
actually breaks the RFC by sending outbound requests on the inbound TLS 
connection. It works, but it’s not RFC compliant.

Cheers,
/O
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Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Olle E. Johansson]

> On 30 Jun 2021, at 09:10, Shahid Hussain  wrote:
> 
> Hi,
> Websocket module documentation has a code reference to use aliases for SIP 
> routing. However, aliases will not work in the following setup and situation.
> 1. Kamailio is configured with active and standby node
> 2. Ping is implemented from webclient and kamailio responds with pong.
> 3. Two clients ClinetA and ClientB registered themselves to Kamailio.
> 4. After SIP negotiation (INVITE-200OK) each client learnt about below 
> aliases.
> Alias of ClientA:
> alias=172.27.6.98~58336
> Alias of ClientB:
> alias=172.27.6.98~58342
>  
> So normally if ClientB wants to send SIP message to ClientA, SIP URI from 
> ClientB looks like below
> ACK sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws 
> SIP/2.0
>  4. Call is in a connected state.
> 
> Following is the issue.
> i.  Switchover (or network lost or reboot) at Kamailio happened
> ii.  Due to ping pong both the clients detected network loss individually and 
> re-registered themselves.
> iii.  Aliases of both the clients got changed.
> New Alias of ClientA:
> alias=172.27.6.98~ 58346
>  
> New Alias of ClientB:
> alias=172.27.6.98~ 58348
>  
> iv.   But ClientB doesn’t know that ClientA also re-registered and ClientA’s 
> alias got changed and vice-versa
> v.Because of this ClientB still sends BYE with Initial alias
> BYE sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws 
> SIP/2.0
> 
> Would like to know what is the recommended solution for this problem using 
> alias or is it a limitation of using alias?

Full implementation of SIP outbound is the only solution close to solving this 
problem in the IETF standards.
However, I have seen no single SIP client that have implemented this, even 
though Kamailio supports
it on the server side. The idea is that you always have two TCP connections to 
two active servers.

https://datatracker.ietf.org/doc/rfc5626/ 


Having said that, SIP outbound enables registration survival and always being 
reachable, but it does not handle dialog survival. It was something we 
discussed for a while at the latest Kamailio dev meeting.

Hopefully the actual RTP streams can survive the failover and the call can go 
on, but if it depends on SIP signalling there’s no other way (that i know of) 
to survive and move the call to a new server with a new TCP connection. I have 
struggled with similar problems and made possible call survival in the case of 
lost SIP signalling path. It requires SIP uas that doesn’t implement any 
keepalive or SIP timer that will hangup if SIP signalling is dead.

/O__
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