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Posted (edited)

good day,

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

rtc_log.txt

Edited by klsheng

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