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LanScape VOIP Media Engine™ - Pre-Sales Technical Support
 LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
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BrunoJuchli
Intermediate
Intermediate


Joined: July 07 2014
Location: Switzerland
Posts: 3
Posted: July 07 2014 at 5:29am | IP Logged Quote BrunoJuchli

Hi

I'm trying to establish a peer to peer call.
The call should be initiated by the VoIP Media Engine, the other peer is a MicroSIP Softphone on the same computer.

The same setup works with most of the competitors products, so i'm sure that MicroSIP is configured correctly.

I've tried dialing 127.0.0.1:54321, sip:127.0.0.1:54321 and sip:foo@127.0.0.1:54321.

127.0.0.1:54321
Adding this address to the phone book results in an error telling me that the address is not valid SIP (that is the expected result, i'm only listing it for completeness` sake)

sip:127.0.0.1:54321
Adding this address to the phone book works.
However, when trying to dial it, the same error occurs as with the above address (invalid format). But not at the same point in the workflow (=inconsistency)
This address format works with most other products, but it may not be standard compliant (i don't know).

sip:foo@127.0.0.1:54321
Adding this address to the phone book works.
Also dialing does not result in an error, but rather it goes on "for ever" = there's no success. With other products an incoming call would pop up @MicroSip.
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BrunoJuchli
Intermediate
Intermediate


Joined: July 07 2014
Location: Switzerland
Posts: 3
Posted: July 07 2014 at 5:37am | IP Logged Quote BrunoJuchli

(I can't find the edit button so i'm adding this reply instead).

I'm using the C# Multiline Phone, and the configuration is left "default", except that i adapted the IP Address and supported Codecs (aLaw instead of uLaw).
Basic --> IP Address: 127.0.0.1
I've also tried with Basic --> "Use "fast dialing"" --> on/off (tried with both)

Sip Proxy/Registrar --> Everything off

I've activated the SIP.log, see below.
Notice that

  • "From" field in the header states IP 29.28.94.92, shouldn't it be 127.0.0.1?

  • o=Anonymous (...) IN IP4 206.197.117.214, again, what IP is this?

  • c=IN IP4 206.197.117.214, again, what IP is this?






************* Log Opened (Jul 07 11:31:31) *************

>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#1, [11:31:39.729] 0 Ms, To: 40.134.247.14:39809) >>>>
INVITE sip:bruno@127.0.0.1:54321 SIP/2.0
Via: SIP/2.0/UDP 83.51.235.241:43911;rport;branch=z9hG4bK00824885
From: "LanScape VOIP" <sip:Anonymous@29.28.94.92:43911>;tag=825093
To: <sip:bruno@127.0.0.1:54321>
Contact: <sip:Anonymous@29.28.94.92:43911>
Call-ID: de862abd-291a-468f-b1fe-d761349afe98-00000acc@127.0.0.1
CSeq: 8559873 INVITE
Max-Forwards: 70
Organization: 9835CAB0-CF6B-4752-9D4E-574ED755B489
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.20 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 233
User-Agent: LanScape VOIP Samples/v1.2.3.4 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=Anonymous 8533972 8533972 IN IP4 206.197.117.214
s=LanScape
c=IN IP4 206.197.117.214
t=0 0
m=audio 41515 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#2, [11:31:41.752] 2023 Ms, To: 40.134.247.14:39809) >>>>
INVITE sip:bruno@127.0.0.1:54321 SIP/2.0
Via: SIP/2.0/UDP 83.51.235.241:43911;rport;branch=z9hG4bK00824885
From: "LanScape VOIP" <sip:Anonymous@29.28.94.92:43911>;tag=825093
To: <sip:bruno@127.0.0.1:54321>
Contact: <sip:Anonymous@29.28.94.92:43911>
Call-ID: de862abd-291a-468f-b1fe-d761349afe98-00000acc@127.0.0.1
CSeq: 8559873 INVITE
Max-Forwards: 70
Organization: 9835CAB0-CF6B-4752-9D4E-574ED755B489
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.20 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 233
User-Agent: LanScape VOIP Samples/v1.2.3.4 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=Anonymous 8533972 8533972 IN IP4 206.197.117.214
s=LanScape
c=IN IP4 206.197.117.214
t=0 0
m=audio 41515 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#3, [11:31:45.754] 4002 Ms, To: 40.134.247.14:39809) >>>>
INVITE sip:bruno@127.0.0.1:54321 SIP/2.0
Via: SIP/2.0/UDP 83.51.235.241:43911;rport;branch=z9hG4bK00824885
From: "LanScape VOIP" <sip:Anonymous@29.28.94.92:43911>;tag=825093
To: <sip:bruno@127.0.0.1:54321>
Contact: <sip:Anonymous@29.28.94.92:43911>
Call-ID: de862abd-291a-468f-b1fe-d761349afe98-00000acc@127.0.0.1
CSeq: 8559873 INVITE
Max-Forwards: 70
Organization: 9835CAB0-CF6B-4752-9D4E-574ED755B489
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.20 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 233
User-Agent: LanScape VOIP Samples/v1.2.3.4 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=Anonymous 8533972 8533972 IN IP4 206.197.117.214
s=LanScape
c=IN IP4 206.197.117.214
t=0 0
m=audio 41515 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#4, [11:31:53.758] 8004 Ms, To: 40.134.247.14:39809) >>>>
INVITE sip:bruno@127.0.0.1:54321 SIP/2.0
Via: SIP/2.0/UDP 83.51.235.241:43911;rport;branch=z9hG4bK00824885
From: "LanScape VOIP" <sip:Anonymous@29.28.94.92:43911>;tag=825093
To: <sip:bruno@127.0.0.1:54321>
Contact: <sip:Anonymous@29.28.94.92:43911>
Call-ID: de862abd-291a-468f-b1fe-d761349afe98-00000acc@127.0.0.1
CSeq: 8559873 INVITE
Max-Forwards: 70
Organization: 9835CAB0-CF6B-4752-9D4E-574ED755B489
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.20 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 233
User-Agent: LanScape VOIP Samples/v1.2.3.4 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=Anonymous 8533972 8533972 IN IP4 206.197.117.214
s=LanScape
c=IN IP4 206.197.117.214
t=0 0
m=audio 41515 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#5, [11:32:09.761] 16003 Ms, To: 40.134.247.14:39809) >>>>
INVITE sip:bruno@127.0.0.1:54321 SIP/2.0
Via: SIP/2.0/UDP 83.51.235.241:43911;rport;branch=z9hG4bK00824885
From: "LanScape VOIP" <sip:Anonymous@29.28.94.92:43911>;tag=825093
To: <sip:bruno@127.0.0.1:54321>
Contact: <sip:Anonymous@29.28.94.92:43911>
Call-ID: de862abd-291a-468f-b1fe-d761349afe98-00000acc@127.0.0.1
CSeq: 8559873 INVITE
Max-Forwards: 70
Organization: 9835CAB0-CF6B-4752-9D4E-574ED755B489
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.20 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 233
User-Agent: LanScape VOIP Samples/v1.2.3.4 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp

v=0
o=Anonymous 8533972 8533972 IN IP4 206.197.117.214
s=LanScape
c=IN IP4 206.197.117.214
t=0 0
m=audio 41515 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=ptime:20
a=fmtp:101 0-16




>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#6, [11:32:20.463] 10702 Ms, To: 207.170.35.49:22466) >>>>
CANCEL sip:bruno@127.0.0.1:54321 SIP/2.0
Via: SIP/2.0/UDP 83.51.235.241:43911;rport;branch=z9hG4bK00824885
From: "LanScape VOIP" <sip:Anonymous@29.28.94.92:43911>;tag=825093
To: <sip:bruno@127.0.0.1:54321>
Call-ID: de862abd-291a-468f-b1fe-d761349afe98-00000acc@127.0.0.1
CSeq: 8559873 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Samples/v1.2.3.4 (www.LanScapeCorp.com)
x-VOIP-SDK: LanScape VOIP Media Engine(Windows)/6.0.1.20 (www.LanScapeCorp.com - This is a trial version not for general deployment)
Content-Length: 0
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BrunoJuchli
Intermediate
Intermediate


Joined: July 07 2014
Location: Switzerland
Posts: 3
Posted: July 07 2014 at 5:48am | IP Logged Quote BrunoJuchli

Ok alrighty i figured it out myself.

I had to replace

status = MediaEngine.MakeCallUri(                         SipUri,                         (AppSettings.UseSipDomain && AppSettings.UseSipProxy),                         PhoneLine,                         false,               // synchronous,                         0                    // timeout Ms.                     );

by

status = MediaEngine.MakeCall("foo", "127.0.0.1", 54321, 0, false, 0);
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