[sip-comm] Why doesn't SIP communicator sending secong INVITE?


#1

I am experiencing the following problem with my SIP network. I have a
server with opensips running and two clients (Owl and Menk), both
behind separate NAT, both NATs with SIP ALG turned ON.

Now I am debugging a SIP protocol exchanging, so no NAT traversal is
configured on server.

I have the following problem:

1) When calling from Menk to Owl, I see initiator ringing and also see
callee ringing. But if I take hook off, initiator continues ringing.
So, it does not feel that calee answered.
2) When calling from Owl to Menk, the situation is even worse -- the
SIP communicator stops at "Initiating Call" message and does not ring.
Menk does not feel somebody calls it.

The problem with the first situation may be interpreted as
impossibility to establish sound communication. So, I postponed it.

The second situation shows strange behavior while network sniffering:

Owl to server:
INVITE sip:user1@company.com SIP/2.0
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
CSeq: 1 INVITE

From: "user2" <sip:user2@company.com>;tag=8fe1cc74

Via: SIP/2.0/UDP
71.58.71.183:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
Max-Forwards: 70
Contact: "user2"
<sip:user2@2.2.3.4:5060;transport=udp;registering_acc=company_com>
User-Agent: SIP Communicator1.0-alpha6-nightly.build.3017Windows XP
Content-Type: application/sdp
Content-Length: 818

Server to Owl:
SIP/2.0 407 Proxy Authentication Required
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
CSeq: 1 INVITE

From: "user2" <sip:user2@company.com>;tag=8fe1cc74

Via: SIP/2.0/UDP
2.2.3.4:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
Proxy-Authenticate: Digest realm="company.com",
nonce="4cbc726e00000017137e7a125622c2fef9bb02d49ae4915e"
Server: OpenSIPS (1.6.2-tls (i386/linux))
Content-Length: 0

Owl to Server:
ACK sip:user1@company.com SIP/2.0
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
Max-Forwards: 70

From: "user2" <sip:user2@company.com>;tag=8fe1cc74

Via: SIP/2.0/UDP
2.2.3.4:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
CSeq: 1 ACK
Content-Length: 0

And that's all. I think a second INVITE is required here. Am I right
or not? If yes, then why SIP communicator can decide not to send it?

Thanks.

···

To: <sip:user1@company.com>
To: <sip:user1@company.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a0a6
To: <sip:user1@company.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a0a6

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#2

Hello Dmitri,

Could you please send us the complete wireshark dump as the server sees it?

Thanks,
Emil

На 18.10.10 18:50, Dmitry Kravchenko написа:

···

I am experiencing the following problem with my SIP network. I have a
server with opensips running and two clients (Owl and Menk), both
behind separate NAT, both NATs with SIP ALG turned ON.

Now I am debugging a SIP protocol exchanging, so no NAT traversal is
configured on server.

I have the following problem:

1) When calling from Menk to Owl, I see initiator ringing and also see
callee ringing. But if I take hook off, initiator continues ringing.
So, it does not feel that calee answered.
2) When calling from Owl to Menk, the situation is even worse -- the
SIP communicator stops at "Initiating Call" message and does not ring.
Menk does not feel somebody calls it.

The problem with the first situation may be interpreted as
impossibility to establish sound communication. So, I postponed it.

The second situation shows strange behavior while network sniffering:

Owl to server:
INVITE sip:user1@company.com SIP/2.0
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
CSeq: 1 INVITE
From: "user2" <sip:user2@company.com>;tag=8fe1cc74
To: <sip:user1@company.com>
Via: SIP/2.0/UDP
71.58.71.183:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
Max-Forwards: 70
Contact: "user2"
<sip:user2@2.2.3.4:5060;transport=udp;registering_acc=company_com>
User-Agent: SIP Communicator1.0-alpha6-nightly.build.3017Windows XP
Content-Type: application/sdp
Content-Length: 818

Server to Owl:
SIP/2.0 407 Proxy Authentication Required
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
CSeq: 1 INVITE
From: "user2" <sip:user2@company.com>;tag=8fe1cc74
To: <sip:user1@company.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a0a6
Via: SIP/2.0/UDP
2.2.3.4:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
Proxy-Authenticate: Digest realm="company.com",
nonce="4cbc726e00000017137e7a125622c2fef9bb02d49ae4915e"
Server: OpenSIPS (1.6.2-tls (i386/linux))
Content-Length: 0

Owl to Server:
ACK sip:user1@company.com SIP/2.0
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
Max-Forwards: 70
From: "user2" <sip:user2@company.com>;tag=8fe1cc74
To: <sip:user1@company.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a0a6
Via: SIP/2.0/UDP
2.2.3.4:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
CSeq: 1 ACK
Content-Length: 0

And that's all. I think a second INVITE is required here. Am I right
or not? If yes, then why SIP communicator can decide not to send it?

Thanks.

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#3

I am sorry, I saw successive invites in dump!

For some reason neither SIP Workbench program, nor opensips itself read them.
A message

/usr/sbin/opensips[10757]: DBG:sl:sl_filter_ACK: local ACK found ->
dropping it!

present while ACK processing. May be this bans successive INVITES. But why?

···

2010/10/18 Emil Ivov <emcho@sip-communicator.org>:

Hello Dmitri,

Could you please send us the complete wireshark dump as the server sees it?

Thanks,
Emil

На 18.10.10 18:50, Dmitry Kravchenko написа:

I am experiencing the following problem with my SIP network. I have a
server with opensips running and two clients (Owl and Menk), both
behind separate NAT, both NATs with SIP ALG turned ON.

Now I am debugging a SIP protocol exchanging, so no NAT traversal is
configured on server.

I have the following problem:

1) When calling from Menk to Owl, I see initiator ringing and also see
callee ringing. But if I take hook off, initiator continues ringing.
So, it does not feel that calee answered.
2) When calling from Owl to Menk, the situation is even worse -- the
SIP communicator stops at "Initiating Call" message and does not ring.
Menk does not feel somebody calls it.

The problem with the first situation may be interpreted as
impossibility to establish sound communication. So, I postponed it.

The second situation shows strange behavior while network sniffering:

Owl to server:
INVITE sip:user1@company.com SIP/2.0
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
CSeq: 1 INVITE
From: "user2" <sip:user2@company.com>;tag=8fe1cc74
To: <sip:user1@company.com>
Via: SIP/2.0/UDP
71.58.71.183:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
Max-Forwards: 70
Contact: "user2"
<sip:user2@2.2.3.4:5060;transport=udp;registering_acc=company_com>
User-Agent: SIP Communicator1.0-alpha6-nightly.build.3017Windows XP
Content-Type: application/sdp
Content-Length: 818

Server to Owl:
SIP/2.0 407 Proxy Authentication Required
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
CSeq: 1 INVITE
From: "user2" <sip:user2@company.com>;tag=8fe1cc74
To: <sip:user1@company.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a0a6
Via: SIP/2.0/UDP
2.2.3.4:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
Proxy-Authenticate: Digest realm="company.com",
nonce="4cbc726e00000017137e7a125622c2fef9bb02d49ae4915e"
Server: OpenSIPS (1.6.2-tls (i386/linux))
Content-Length: 0

Owl to Server:
ACK sip:user1@company.com SIP/2.0
Call-ID: f53bda61becfbd266bd51718dca31667@0.0.0.0
Max-Forwards: 70
From: "user2" <sip:user2@company.com>;tag=8fe1cc74
To: <sip:user1@company.com>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a0a6
Via: SIP/2.0/UDP
2.2.3.4:5060;branch=z9hG4bKb80be33d98edf0204cafbe8db5eabb20313236
CSeq: 1 ACK
Content-Length: 0

And that's all. I think a second INVITE is required here. Am I right
or not? If yes, then why SIP communicator can decide not to send it?

Thanks.

---------------------------------------------------------------------
To unsubscribe, e-mail: users-unsubscribe@sip-communicator.dev.java.net
For additional commands, e-mail: users-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31