From SIP to RTP (Part 6) – The phone is ringing….

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Real Case: Asterisk receive an external call
A call from an external number to our pbx using a SIP trunk.
Att: I have “sniffed” that traffic using tcpdump.

tcpdump -i <interface> -s 65535 -w <file name>

In the next the Asterisk pbx is inside a LAN network, and its ip address is 10.10.10.110. The router in the network is configured with a public ip address 79.14.212.52. Asterisk is configured correctly using local network & public network parameters: in this mode all the message in SIP will be correctly valuated.

File sip.conf

[general]
externip = 79.14.212.52
localnet=10.10.10.0/255.255.255.0

>>Message from ISTP provider (IP 212.97.59.76) to Pbx (ip: 10.10.10.110)

INVITE sip:5224851@79.14.212.52 SIP/2.0
Record-Route: <sip:212.97.59.76:5061;lr=on;ftag=as16df4853;rpp=np>
Via: SIP/2.0/UDP 212.97.59.76:5061;branch=z9hG4bKc7f1.f1fb7516.1
Via: SIP/2.0/UDP 212.97.59.85:5060;branch=z9hG4bK683ee34f;rport=5060
From: "+393461050897" <sip:+393461050897@sip.messagenet.it>;tag=as16df4853
To: <sip:01042064023@212.97.59.76:5061>
Contact: <sip:+393461050897@212.97.59.85>
Call-ID: 509017dd3847bf900f839d877abfd752@sip.messagenet.it
CSeq: 102 INVITE
User-Agent: foxtrot
Max-Forwards: 69
Date: Tue, 29 Nov 2011 00:29:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 21598 21598 IN IP4 193.227.104.40
s=session
c=IN IP4 193.227.104.40
t=0 0
m=audio 35936 RTP/AVP 18 3 97 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

The trace show an incoming call from number +393461050897 to our number (trunk with ITSP) that is 01042064023.

Att.: 5224851 is another internal number assigned from ITSP to the trunk that is linked to 01042064023: very commonly the ITSP use this “internal number” for REGISTER & OPTIONS messages.

Att.: The ITSP know where is located the our pbx because Asterisk send periodically a REGISTER & OPTIONS messages: the NAT is “opened” by this kind of messages.

The ITSP wants to receive the voice streaming to 193.227.104.40 port 35936.

Att: Very interesting the UA that receive the voice stream is different from the UA that establish the call (different ip address): it is very common using big ITSP.

>> Messages from PBX (ip 10.10.10.110) to ISTP provider (IP 212.97.59.76).

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.97.59.76:5061;branch=z9hG4bKc7f1.f1fb7516.1;received=212.97.59.76
Via: SIP/2.0/UDP 212.97.59.85:5060;branch=z9hG4bK683ee34f;rport=5060
Record-Route: <sip:212.97.59.76:5061;lr=on;ftag=as16df4853;rpp=np>
From: "+393461050897" <sip:+393461050897@sip.messagenet.it>;tag=as16df4853
To: <sip:01042064023@212.97.59.76:5061>
Call-ID: 509017dd3847bf900f839d877abfd752@sip.messagenet.it
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.4.40)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:5224851@79.14.212.52>
Content-Length: 0

>> Messages from PBX (ip 10.10.10.110) to ISTP provider (IP 212.97.59.76).

SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.97.59.76:5061;branch=z9hG4bKc7f1.f1fb7516.1;received=212.97.59.76
Via: SIP/2.0/UDP 212.97.59.85:5060;branch=z9hG4bK683ee34f;rport=5060
Record-Route: <sip:212.97.59.76:5061;lr=on;ftag=as16df4853;rpp=np>
From: "+393461050897" <sip:+393461050897@sip.messagenet.it>;tag=as16df4853
To: <sip:01042064023@212.97.59.76:5061>;tag=as5dfb4137
Call-ID: 509017dd3847bf900f839d877abfd752@sip.messagenet.it
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.4.40)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:5224851@79.14.212.52>
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 2754 2754 IN IP4 79.14.212.52
s=session
c=IN IP4 79.14.212.52
t=0 0
m=audio 18480 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

>>Final ACK

ACK sip:5224851@79.14.212.52 SIP/2.0
Record-Route: <sip:212.97.59.76:5061;lr=on;ftag=as16df4853>
Via: SIP/2.0/UDP 212.97.59.76:5061;branch=0
Via: SIP/2.0/UDP 212.97.59.85:5060;branch=z9hG4bK0bbcf9cd;rport=5060
From: "+393461050897" <sip:+393461050897@sip.messagenet.it>;tag=as16df4853
To: <sip:01042064023@212.97.59.76:5061>;tag=as5dfb4137
Contact: <sip:+393461050897@212.97.59.85>
Call-ID: 509017dd3847bf900f839d877abfd752@sip.messagenet.it
CSeq: 102 ACK
User-Agent: foxtrot
Max-Forwards: 69
Content-Length: 0

Asterisk accepts the call and declare that want to receive voice stream to 79.14.212.52 port 18480.

Now start the voice streaming: in this figure this traffic “sniffed” and showed using wireshark.

PBX → ITSP
10.10.10.110 – src port 1840 → 193.227.104.40 dst port 35936

ITSP → PBX
193.227.104.40 src port 35936 → 10.10.10.110 – dst port 1840

We can see that Asterisk start to send packet: in this mode the NAT open the port correctly and the ITSP can reach the Asterisk PBX inside the LAN.

Another very interesting things is that tha ITSP UA use a different Ip Address for the SIP messages and for the RTP stream.

Antother Real Case: Interconnection Asterisk<->Avaya/Nortel BCM 450
In this case an external connected to an Avaya Pbx BCM450 call using SIP an Asterisk PBX.

Asterisk PBX – IP 10.10.10.110
Avaya PBX – IP 10.10.10.155
Extent connected to Avaya PBX that establish the call – IP 10.10.10.87

>> From Avaya to Asterisk

INVITE sip:420;phone-context=subscriber.private@10.10.10.110:5060;transport=udp;user=phone SIP/2.0
From: <sip:anonymous@anonymous.invalid>;tag=34e03fa8-a0a0a9b-13c4-55013-59a-5708f3a4-59a
To: <sip:420;phone-context=subscriber.private@10.10.10.110;user=phone>
Call-ID: 35036840-a0a0a9b-13c4-55013-59a-3609da9e-59a
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.10.10.155:5060;branch=z9hG4bK-59a-15e419-1faa1b5d
Max-Forwards: 70
Supported: sipvc,x-nortel-sipvc,100rel,replaces
User-Agent: Nortel Networks BCM VoIP Gateway release_46 version_46.46.0.33
P-Asserted-Identity: <sip:anonymous@10.10.10.155>
Privacy: id;user
x-nt-corr-id: 35036840-a0a0a9b-13c4-55013-59a-3609da9e-59a
Contact: <sip:anonymous@anonymous.invalid:5060;maddr=10.10.10.155;transport=udp>
Allow: INVITE,INFO,ACK,OPTIONS,CANCEL,BYE,NOTIFY,PRACK,UPDATE,REFER
Content-Type: application/sdp
Content-Length: 334
v=0
o=- 1323100317 1323100317 IN IP4 10.10.10.155
s=-
c=IN IP4 10.10.10.87
t=0 0
a=sqn:0
a=cdsc:1 image udptl t38
m=audio 51000 RTP/AVP 18 4 0 8 120 111
c=IN IP4 10.10.10.87
a=fmtp:18 annexb=yes
a=fmtp:4 annexa=yes
a=rtpmap:120 telephone-event/8000
a=fmtp:120 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:30
a=sendrecv

Asterisk -> Avaya

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.10.10.155:5060;branch=z9hG4bK-59a-15e419-1faa1b5d;received=10.10.10.155
From: <sip:anonymous@anonymous.invalid>;tag=34e03fa8-a0a0a9b-13c4-55013-59a-5708f3a4-59a
To: <sip:420;phone-context=subscriber.private@10.10.10.110;user=phone>
Call-ID: 35036840-a0a0a9b-13c4-55013-59a-3609da9e-59a
CSeq: 1 INVITE
User-Agent: FPBX-2.8.1(1.4.40)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:420@10.10.10.110>
Content-Length: 0

From Asterisk to Avaya

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.155:5060;branch=z9hG4bK-59a-15e419-1faa1b5d;received=10.10.10.155
From: <sip:anonymous@anonymous.invalid>;tag=34e03fa8-a0a0a9b-13c4-55013-59a-5708f3a4-59a
To: <sip:420;phone-context=subscriber.private@10.10.10.110;user=phone>;tag=as5b7d8e50
Call-ID: 35036840-a0a0a9b-13c4-55013-59a-3609da9e-59a
CSeq: 1 INVITE
User-Agent: FPBX-2.8.1(1.4.40)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:420@10.10.10.110>
Content-Type: application/sdp
Content-Length: 206
v=0
o=root 2756 2756 IN IP4 10.10.10.110
s=session
c=IN IP4 10.10.10.110
t=0 0
m=audio 16316 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

It is possible to verify with this Sip packet that the BCM 450 does not behave like a B2BUA: the asterisk pbx & extent of Avaya/Nortel BCM 450 connect itself each other directly !

PREVIOUS POST: From Sip to RTP (Part 5) – Trunks & surroundings

Linkografia
http://www.ietf.org/rfc/rfc3325.txt
http://en.wikipedia.org/wiki/Session_Initiation_Protocol

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