Hello
I'm trying to integrate Nokia phones (e61i ; e65 ; e51 ; e90) with a VOIP Architecture.
There are no problems with outbounds calls.
We are experimenting problems with inbounds calls:
[SIZE="6"]FIRST PROBLEM :[/SIZE]
When the sip signalization begin the phone receive a first INVITE.There is no SDP information in this INVITE.
Then the SIP Proxy send a RE-INVITE with SDP information but the phone do not seems to catch these packet. We think that the Nokia SIP Stack does not recognize RE-INVITE packets. In our Ethereal/Wireshark capture file the RE-INVITE looks to be interpreted as an new INVITE packet by the phone.
This problem is not present with the E65 with 1.0633.18.01 firmware. Unfortunately we upgraded the firmware on this phone.
Is that a known problem with the SIP Stack of Nokia phones ? Or have you any information on this subject ?
Here is my SIP/RTP call flow if you want see the problem in details.
-------->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: [email][email protected][/email]
Contact: <sip:yyy.yy.yyy.yyy:5060>
CSeq: 6180794 INVITE
From: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Max-Forwards: 31
P-Asserted-Identity: <sip:[email protected];user=phone>
P-Preferred-Identity: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>
User-Agent: SIPProxy/v4.41f (gw_sip)
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5FB8-3D480B
Content-Length: 0
<--------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5FB8-3D480B
To: <sip:[email protected];user=phone>
From: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Call-ID: [email][email protected][/email]
CSeq: 6180794 INVITE
Content-Length: 0
<--------
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5FB8-3D480B
To: <sip:[email protected];user=phone>;tag=eaus790736q7ceueldg5b432
Contact: <sip:[email protected]:5060;transport=UDP>
From: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Call-ID: [email][email protected][/email]
CSeq: 6180794 INVITE
Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 439
v=0
o=Nokia-SIPUA 63382497070157250 63382497070157250 IN IP4 xx.x.x.xx
s=-
c=IN IP4 xx.x.x.xx
t=0 0
m=audio 49152 RTP/AVP 96 0 8 97 18 98 13
a=sendrecv
a=ptime:20
a=maxptime:200
a=fmtp:96 mode-change-neighbor=1
a=fmtp:18 annexb=no
a=fmtp:98 0-15
a=rtpmap:96 AMR/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:97 iLBC/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:98 telephone-event/8000/1
a=rtpmap:13 CN/8000/1
-------->
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email][email protected][/email]
Contact: <sip:yyy.yy.yyy.yyy:5060>
Content-Type: application/sdp
CSeq: 6180794 ACK
From: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Max-Forwards: 31
To: <sip:[email protected];user=phone>;tag=eaus790736q7ceueldg5b432
User-Agent: SIPProxy/v4.41f (gw_sip)
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-475C-3D4834
Content-Length: 280
v=0
o=cp10 121423391582 121423391582 IN IP4 0.0.0.0
s=SIP Call
c=IN IP4 172.0.0.0
t=0 0
m=audio 65534 RTP/AVP 18 8 101
b=AS:64
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
-------->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: [email][email protected][/email]
Contact: <sip:yyy.yy.yyy.yyy:5060>
Content-Type: application/sdp
CSeq: 6180795 INVITE
From: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Max-Forwards: 31
P-Asserted-Identity: <sip:[email protected];user=phone>
P-Preferred-Identity: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>;tag=eaus790736q7ceueldg5b432
User-Agent: SIPProxy/v4.41f (gw_sip)
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5CDD-3D4836
Content-Length: 284
v=0
o=cp10 121423391582 121423391585 IN IP4 37.5.8.86
s=SIP Call
c=IN IP4 212.39.140.113
t=0 0
m=audio 34582 RTP/AVP 18 8 98
b=AS:64
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-15
a=ptime:30
a=sendrecv
<--------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-5CDD-3D4836
To: <sip:[email protected];user=phone>;tag=eaus790736q7ceueldg5b432
From: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Call-ID: [email][email protected][/email]
CSeq: 6180795 INVITE
Content-Length: 0
====================
UNDIRECTIONNAL RTP WHITE COMMUNICATION (NO VOICE FLOW)
====================
<--------
BYE sip:yyy.yy.yyy.yyy:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xx.x.x.xx:5060;branch=z9hG4bK4rmqmo491dhc7pa399t7kqm;rport
To: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
From: <sip:[email protected];user=phone>;tag=eaus790736q7ceueldg5b432
Call-ID: [email][email protected][/email]
CSeq: 1 BYE
Max-Forwards: 70
Content-Length: 0
-------->
SIP/2.0 200 OK
Call-ID: [email][email protected][/email]
CSeq: 1 BYE
From: <sip:[email protected];user=phone>;tag=eaus790736q7ceueldg5b432
Server: SIPProxy/v4.41f (gw_sip)
To: " " <sip:[email protected];user=phone>;tag=22708-AN-006ef424-711b880c6
Via: SIP/2.0/UDP xx.x.x.xx:5060;received=xx.x.x.xx;rport=5060;branch=z9hG4bK4rmqmo491dhc7pa399t7kqm
Content-Length: 0
[SIZE="6"]SECOND PROBLEM :
[/SIZE]
We had a chance to have a Nokia E61i phone with firmware 1.0633.22.5
With this firmware the sip captures are the same, but we do not have the problem describe below.
1 - All calls from an MGCP phone are working.
2 - Calls from mobile phone or from analogic phone are not working everytime.
In almost 25% of calls we are experimtings some problems. There is no problem in signalisation (all the same)but calls duration is around 1 second and the caller phone hang-up automaticly.The mobile phone notice the user with a message like "Service unavailable". .
We have analyzed the SIP Flow, and noticed a strange comportement.
While the codec negociation, the Nokia Phone do not replay with any codecs and the SIP Proxy answer by a bye.
Here is the SIP Flow if you can help me :
----->
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: [email][email protected][/email]
Contact: <sip:yyy.yy.yyy.yyy:5060>
Content-Type: application/sdp
CSeq: 7159120 INVITE
From: "caller_number" <sip:[email protected];user=phone>;tag=11282-BI-007fea8a-2a619f720
Max-Forwards: 31
P-Access-Network-Info: ADSL;dsl_location="NOA=3;APRI=1;ADD=627613000";network-provided
P-Asserted-Identity: <sip:[email protected];user=phone>
To: <sip:[email protected];user=phone>;tag=062ut1cf22gbikcs2i7uokr2
User-Agent: SIPProxy/v4.41f (gw_sip)
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-DD8-45B7B4
Content-Length: 234
v=0
o=cp10 121448350346 121448350349 IN IP4 212.39.140.5
s=SIP Call
c=IN IP4 212.39.140.101
t=0 0
m=audio 32250 RTP/AVP 8 18
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
<--------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-DD8-45B7B4
To: <sip:[email protected];user=phone>;tag=062ut1cf22gbikcs2i7uokr2
From: "caller_number" <sip:[email protected];user=phone>;tag=11282-BI-007fea8a-2a619f720
Call-ID: [email][email protected][/email]
CSeq: 7159120 INVITE
Content-Length: 0
<--------
SIP/2.0 200 OK
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-DD8-45B7B4
To: <sip:[email protected];user=phone>;tag=062ut1cf22gbikcs2i7uokr2
Contact: <sip:[email protected]:5060;transport=UDP>
From: "caller_number" <sip:[email protected];user=phone>;tag=11282-BI-007fea8a-2a619f720
Call-ID: [email][email protected][/email]
CSeq: 7159120 INVITE
Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 182
v=0
o=Nokia-SIPUA 63335907022772750 63335907022772751 IN IP4 xx.x.x.xx
s=-
c=IN IP4 xx.x.x.xx
t=0 0
m=audio 0 RTP/AVP 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000/1
------>
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email][email protected][/email]
Contact: <sip:yyy.yy.yyy.yyy:5060>
CSeq: 7159120 ACK
From: "caller_number" <sip:[email protected];user=phone>;tag=11282-BI-007fea8a-2a619f720
Max-Forwards: 31
To: <sip:[email protected];user=phone>;tag=062ut1cf22gbikcs2i7uokr2
User-Agent: SIPProxy/v4.41f (gw_sip)
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-2EEF-45B7B6
Content-Length: 0
------->
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Call-ID: [email][email protected][/email]
CSeq: 7159121 BYE
From: "caller_number" <sip:[email protected];user=phone>;tag=11282-BI-007fea8a-2a619f720
Max-Forwards: 31
Reason: q.850;cause=65
To: <sip:[email protected];user=phone>;tag=062ut1cf22gbikcs2i7uokr2
User-Agent: SIPProxy/v4.41f (gw_sip)
Via: SIP/2.0/UDP yyy.yy.yyy.yyy:5060;branch=z9hG4bK-1E3C-45B7B7
Content-Length: 0
Thanks if you can help.