Quote:
Originally Posted by martin
The record-route processing has now been removed from SIP Broker. Please confirm that the new SIP Broker call handling is working as requested.
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Thank you for that, Martin. I'm humbled that my suggestion has been implemented.
Here are the SIP INVITE messages that arrived at my SIP phone when calling via SIP Broker using various methods:
(My PSTN number +8150xxxxxxxx is mapped to the SIP URI
050xxxxxxxx@221.xx.xx.yyy at e164.org)
1. Ordinary SIP call to 8150xxxxxxxx@sipbroker.com
(dialing from my Atcom AT-530 at 221.xx.xx.xx)
RECEIVE << 64.34.162.221:5060 ( 64.34.162.221 = sipbroker.com )
Code:
INVITE sip:050xxxxxxxx@221.xx.xx.xx.yyy SIP/2.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bKaf6.6f14e895.0
Via: SIP/2.0/UDP 221.xx.xx.xx.xx:6080;branch=zh4K088716000
From: 6173xxxxxxx <sip:6173xxxxxxx@sip.pennytel.com>;tag=2267814189
To: "8150xxxxxxxx" <sip:8150xxxxxxxx@sipbroker.com:5060>
Call-ID: 07108322702@221.xx.xx.xx.xx
CSeq: 1 INVITE
Contact: <sip:6173xxxxxxx@221.xx.xx.xx.xx:6080>
max-forwards: 69
supported: replaces, 100rel
user-agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 342
v=0
o=sdp_admin 28694149 31133503 IN IP4 221.xx.xx.xx.xx
s=A conversation
c=IN IP4 221.xx.xx.xx.xx
t=0 0
m=audio 10026 RTP/AVP 0 4 4 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723high/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
2. Calling via the Brisbane (Australia) SIP Broker PSTN gateway ( +61 7 3018 2881 )
RECEIVE << 64.34.162.221:5060
Code:
INVITE sip:050xxxxxxxx@221.xx.xx.xx.yyy SIP/2.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bK0e1e.9be19477.0
Via: SIP/2.0/UDP 204.11.194.10:5070;branch=z9hG4bK10d4f0a6
From: "6173xxxxxxx" <sip:6173xxxxxxx@204.11.194.10:5070>;tag=as3707e432
To: <sip:*0138150xxxxxxxx#@sipbroker.com>
Contact: <sip:6173xxxxxxx@204.11.194.10:5070>
Call-ID: e1cce9171ea3df@204.11.194.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Sun, 29 Jul 2007 13:42:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 393
v=0
o=root 25390 25390 IN IP4 204.11.194.10
s=session
c=IN IP4 204.11.194.10
t=0 0
m=audio 18512 RTP/AVP 0 18 8 111 97 3 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(note the large number of codecs supported by this PSTN gateway)
3. “spiral” test, via Keiichi HONDA's SIP-IX service.
Caller -> sipix.jp -> sipbroker.com -> sipix.jp -> sipbroker.com -> destination
( 58.158.169.93 = sipix.jp )
RECEIVE << 64.34.162.221:5060
Code:
INVITE sip:050xxxxxxxx@221.xx.xx.xx.yyy SIP/2.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bKc62d.99f92f84.0
Via: SIP/2.0/UDP 58.158.169.93;branch=z9hG4bKc62d.b740e301.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bKc62d.89f92f84.0
Via: SIP/2.0/UDP 58.158.169.93;branch=z9hG4bKc62d.a740e301.0
Via: SIP/2.0/UDP 221.xx.xx.xx.xx:6080;branch=z9hG4bK463813099308327117
From: 6173xxxxxxx <sip:6173xxxxxxx@sip.pennytel.com>;tag=2254816291
To: "0101*809601018150xxxxxxxx" <sip:0101*809601018150xxxxxxxx@sipix.jp:5060>
Call-ID: 421904227@221.xx.xx.xx.xx
CSeq: 1 INVITE
Contact: <sip:6173xxxxxxx@221.xx.xx.xx.xx:6080>
max-forwards: 13
supported: replaces, 100rel
user-agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 341
v=0
o=sdp_admin 14392964 7747828 IN IP4 221.xx.xx.xx.xx
s=A conversation
c=IN IP4 221.xx.xx.xx.xx
t=0 0
m=audio 10034 RTP/AVP 0 4 4 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723high/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
4. Using Web Callback (WebCall), as explained here:
http://forum.voxalot.com/e164-org-su...html#post10967
RECEIVE << 64.34.162.221:5060
Code:
INVITE sip:050xxxxxxxx@221.xx.xx.yyy SIP/2.0
Via: SIP/2.0/UDP 64.34.162.221;branch=z9hG4bKef2e.ac94e9e1.0
Via: SIP/2.0/UDP 202.60.75.46:5061;branch=z9hG4bK058f6bf3;rport=5061
From: "Voxalot" <sip:82xxxx@202.60.75.46:5061>;tag=as7b280933
To: <sip:*0138150xxxxxxxx@sipbroker.com>
Contact: <sip:82xxxx@202.60.75.46:5061>
Call-ID: 58b13e4741a6167018277cbe11707d72@202.60.75.46
CSeq: 102 INVITE
User-Agent: voxaLot
Max-Forwards: 69
Date: Sun, 29 Jul 2007 14:11:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
P-src-ip: 221.xx.xx.xx.yyy
Content-Type: application/sdp
Content-Length: 336
v=0
o=root 19704 19704 IN IP4 202.60.75.46
s=session
c=IN IP4 202.60.75.46
t=0 0
m=audio 15482 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
( 202.60.75.46 = au.voxalot.com )
( 82xxxx = six-digit VoXaLot number )
The “Record-Route:” header, together with “lr=on” is now gone from the calls routed via SIP Broker.
All of the above calls should now work with SIP devices that are fussy about “lr” / “lr=on”, including the NTT-West SVIII ADSL modem, which I was using last year.
I think this is an important improvement in interoperability. The SVIII modem is representative of the most widely used unlocked SIP ATA in Japan – those provided for use with the bundled “IP Phone” services provided by Japanese ISPs using NTT-West / NTT-East “Flets” ADSL connection. I intend to write in more detail about this at a later date, because supporting these SIP devices would be, at this point in time, the most effective way to drive adoption of ENUM and VoIP peering among Japanese VoIP users.
As a final example, number (5) doesn't use the sipbroker.com SIP proxy – VoXaLot does the ENUM lookup and routes the call (note the different IP address from which the SIP INVITE arrived).
5. Using VoXaLot (ENUM match in dial plan)
RECEIVE << 64.34.173.199:5060 ( 64.34.173.199 = us.voxalot.com )
Code:
INVITE sip:050xxxxxxxx@221.xx.xx.xx.yyy SIP/2.0
Record-Route: <sip:64.34.173.199;lr=on;ftag=66638464>
Via: SIP/2.0/UDP 64.34.173.199;branch=z9hG4bKd10a.9fba7631.0
Via: SIP/2.0/UDP 221.xx.xx.xx.xx:6080;branch=z9hG4bK2746116113766423988
From: 82xxxx <sip:82xxxx@us.voxalot.com>;tag=66638464
To: "8150xxxxxxxx" <sip:8150xxxxxxxx@us.voxalot.com>
Call-ID: 2180323145-342329725256@221.xx.xx.xx.xx
CSeq: 2 INVITE
Contact: <sip:82xxxx@221.xx.xx.xx.xx:6080>
max-forwards: 69
supported: replaces, 100rel
user-agent: Voip Phone 1.0
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 342
P-hint: outbound
P-src-ip: 221.xx.xx.xx.xx
v=0
o=sdp_admin 31805841 18160206 IN IP4 221.xx.xx.xx.xx
s=A conversation
c=IN IP4 221.xx.xx.xx.xx
t=0 0
m=audio 10036 RTP/AVP 0 4 4 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723high/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
This call would be ignored by the ATA built into the NTT-West SVIII ADSL modem (with the current firmware), because of the “=on” part of the “lr” parameter in the “Record-Route:” header.
Edited:
I don't have the SVIII modem myself anymore, but I know some people who do, so I'll test that when I can.