i have problem in RTC Client 1.3 HOLD control
when using softphone(rtc) with softphone(rtc), i cannot hold the call
but when using softphone(rtc) with sipphone/ip phone , i can hold the call
but once unhold the call, both side cannot hear the voice
i check the sip message, i found that the port number has been changed,
why port number will change? is there any solution so that both side can hear voice after unhold?
i test samples provided by ms, they can hold the call, but those application is direct ip to ip call, and the sip message also shown that the port number has been changed.
does anyone involve in RTC Client development before and face this problem also?
call 01548408907:
INVITE sip:01548408907@218.189.19.37 SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 1 INVITE
Contact: <sip:192.228.118.173:11564>
User-Agent: RTC/1.3
Content-Type: application/sdp
Content-Length: 291
v=0
o=- 0 0 IN IP4 192.228.118.173
s=session
c=IN IP4 192.228.118.173
b=CT:1000
t=0 0
m=audio 16384 RTP/AVP 97 0 8 4 101
a=rtpmap:97 red/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:rejected
trying
100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.0 (i386/linux))
Content-Length: 0
Warning: 392 218.189.19.37:5060 "Noisy feedback tells: pid=19331 req_src_ip=219.94.42.174 req_src_port=2960 in_uri=sip:01548408907@218.189.19.37 out_uri=sip:01548408907@219.94.42.174:5060 via_cnt==1"
ringing
180 Ringing
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on>
Contact: <sip:01548408907@192.228.118.223:5060>
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 1 INVITE
Content-Length: 0
accept call
200 OK
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on>
Contact: <sip:01548408907@219.94.42.174:5060>
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 265
v=0
o=01548408907 514117283 514117283 IN IP4 192.228.118.223
s=CrystalMedia Session
c=IN IP4 218.189.19.37
t=0 0
m=audio 36456 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:60001
a=direction:both
a=sendrecv
ack from me
ACK sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 1 ACK
Route: <sip:01548408907@219.94.42.174:5060>
User-Agent: RTC/1.3
Content-Length: 0
i hold the call now
?INVITE sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 2 INVITE
Route: <sip:01548408907@219.94.42.174:5060>
Contact: <sip:192.228.118.173:11564>
User-Agent: RTC/1.3
Content-Type: application/sdp
Content-Length: 168
v=0
o=- 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
trying
100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 2 INVITE
Server: Sip EXpress router (0.9.0 (i386/linux))
Content-Length: 0
Warning: 392 218.189.19.37:5060 "Noisy feedback tells: pid=19326 req_src_ip=219.94.42.174 req_src_port=2960 in_uri=sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on out_uri=sip:01548408907@219.94.42.174:5060 via_cnt==1"
success hold call
200 OK
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on>
Contact: <sip:01548408907@219.94.42.174:5060>
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 2 INVITE
Content-Type: application/sdp
Content-Length: 173
v=0
o=01548408907 1442463500 1442463500 IN IP4 192.228.118.223
s=CrystalMedia Session
c=IN IP4 0.0.0.0
t=0 0
m=audio 36456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=recvonly
ack
ACK sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 2 ACK
Route: <sip:01548408907@219.94.42.174:5060>
User-Agent: RTC/1.3
Content-Length: 0
unhold call now
INVITE sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 3 INVITE
Route: <sip:01548408907@219.94.42.174:5060>
Contact: <sip:192.228.118.173:11564>
User-Agent: RTC/1.3
Content-Type: application/sdp
Content-Length: 195
v=0
o=- 0 0 IN IP4 192.228.118.173
s=session
c=IN IP4 192.228.118.173
b=CT:1000
t=0 0
m=audio 34046 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
trying
100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 3 INVITE
Server: Sip EXpress router (0.9.0 (i386/linux))
Content-Length: 0
Warning: 392 218.189.19.37:5060 "Noisy feedback tells: pid=19326 req_src_ip=219.94.42.174 req_src_port=2960 in_uri=sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on out_uri=sip:01548408907@219.94.42.174:5060 via_cnt==1"
success unhold call
200 OK
Via: SIP/2.0/UDP 192.228.118.173:11564;rport=2960;received=219.94.42.174
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Record-Route: <sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on>
Contact: <sip:01548408907@219.94.42.174:5060>
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 3 INVITE
Content-Type: application/sdp
Content-Length: 209
v=0
o=01548408907 541153517 541153517 IN IP4 192.228.118.223
s=CrystalMedia Session
c=IN IP4 218.189.19.37
t=0 0
m=audio 36456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=rtcp:60001
a=direction:both
a=sendrecv
ack from me
ACK sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 3 ACK
Route: <sip:01548408907@219.94.42.174:5060>
User-Agent: RTC/1.3
Content-Length: 0
bye message
BYE sip:218.189.19.37;ftag=587858c1884f4ab0a98cc0a5a3222179;lr=on SIP/2.0
Via: SIP/2.0/UDP 192.228.118.173:11564
Max-Forwards: 70
From: "klsheng" <sip:01548408906@218.189.19.37>;tag=587858c1884f4ab0a98cc0a5a3222179;epid=5e1c16dd84
To: <sip:01548408907@218.189.19.37>;tag=6489fa9e
Call-ID: 938d582d1cec4493a7a4dadcf688945d
CSeq: 4 BYE
Route: <sip:01548408907@219.94.42.174:5060>
User-Agent: RTC/1.3
Content-Length: 0
...............
these are the sip message
can someone please figure out for me
why both side cannot hear the voice after unhold call
second question is about CoCreateInstance
if i have 2 dll files
both also use CoInitializeEx
will the 1st dll file affect the 2nd one?
because i always CoCreateInstance fails
i try on 2 application, one is onli 1 dll,
another one is more than one, some of them may b using CoInitializeEx function
the 1st application can work
but the 2nd one cannot work
does anybody know this problem and teach me solve this problem?
thanks a lot
the attachment file is txt file
but the original file is saved from ethereal
Attached File(s)
-
rtc_log.txt (10.76K)
Number of downloads: 18
This post has been edited by klsheng: 14 September 2005 - 03:23 AM



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