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Incorret fmpt of telephone-event in received INVITE
How to connect asterisk to skype using siptosis?is SDP sendonly means to open one RTP Audio stream in this case?Mobicents - RestComm IssueAAC SDP Payload for receive only streamHuawei 9000 HD Video TerminalWebRTC muxing using BUNDLE in JavaSIP codec negotiationWhat's missing in Answer SDP (From web browser to android device)Error 488 Not acceptable hereWebrtc SDP Audio is not working on sender side but receiver can listen sender voice
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We have a case where we have to apply the conditional codec policy on the egress side. But I have a problem with the script where sending telephone-event with payload-type 101 and 119 in the initial INVITE but did not received any fmtp for telephone-event whereas my scripts expects it to come. I am new to this field of SIP and SDP and unable to figure out the exact problem.
I thought the script is expecting what it shouldn't, so removed the expectation and the call is successfully completing. Below is the Sending and Received INVITE.
Sending INVITE with below SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.193
s=-
c=IN IP4 192.168.205.193
t=0 0
m=audio 10000 RTP/AVP 96 97 119
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:119 telephone-event/8000
a=fmtp:97 octet-align=1
Received INVITE with SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.195
s=-
c=IN IP4 192.168.205.195
t=0 0
m=audio 13008 RTP/AVP 102 100 0 96 97 101 119
a=rtpmap:102 AMR-WB/16000/1
a=fmtp:102 mode-set=0,1,2
a=rtpmap:100 AMR/8000
a=fmtp:100 mode-set=0,2,5,7
a=rtpmap:0 PCMU/8000
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=fmtp:97 octet-align=1
a=rtpmap:101 telephone-event/16000
a=rtpmap:119 telephone-event/8000
My script is expecting fmtp: 101 0-15 but missing from received INVITE, When and in which case fmtp of DTMF should be expected and with what payload type of dynamic codec should we receive? What if I remove the fmtp expectation of telephone-event in the received INVITE from the script?
sip payload sdp
asked Mar 24 at 10:46
AnuAnu
459
add a comment |
We have a case where we have to apply the conditional codec policy on the egress side. But I have a problem with the script where sending telephone-event with payload-type 101 and 119 in the initial INVITE but did not received any fmtp for telephone-event whereas my scripts expects it to come. I am new to this field of SIP and SDP and unable to figure out the exact problem.
I thought the script is expecting what it shouldn't, so removed the expectation and the call is successfully completing. Below is the Sending and Received INVITE.
Sending INVITE with below SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.193
s=-
c=IN IP4 192.168.205.193
t=0 0
m=audio 10000 RTP/AVP 96 97 119
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:119 telephone-event/8000
a=fmtp:97 octet-align=1
Received INVITE with SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.195
s=-
c=IN IP4 192.168.205.195
t=0 0
m=audio 13008 RTP/AVP 102 100 0 96 97 101 119
a=rtpmap:102 AMR-WB/16000/1
a=fmtp:102 mode-set=0,1,2
a=rtpmap:100 AMR/8000
a=fmtp:100 mode-set=0,2,5,7
a=rtpmap:0 PCMU/8000
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=fmtp:97 octet-align=1
a=rtpmap:101 telephone-event/16000
a=rtpmap:119 telephone-event/8000
My script is expecting fmtp: 101 0-15 but missing from received INVITE, When and in which case fmtp of DTMF should be expected and with what payload type of dynamic codec should we receive? What if I remove the fmtp expectation of telephone-event in the received INVITE from the script?
sip payload sdp
asked Mar 24 at 10:46
AnuAnu
459
Please edit your question to make it clear: 1/ what side are you implementing? 2/ make a comparison between working scenario & non-working scenario? 3/ And finally, give an exact list of questions in your question...?
– AymericM
Mar 24 at 12:43
add a comment |
We have a case where we have to apply the conditional codec policy on the egress side. But I have a problem with the script where sending telephone-event with payload-type 101 and 119 in the initial INVITE but did not received any fmtp for telephone-event whereas my scripts expects it to come. I am new to this field of SIP and SDP and unable to figure out the exact problem.
I thought the script is expecting what it shouldn't, so removed the expectation and the call is successfully completing. Below is the Sending and Received INVITE.
Sending INVITE with below SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.193
s=-
c=IN IP4 192.168.205.193
t=0 0
m=audio 10000 RTP/AVP 96 97 119
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:119 telephone-event/8000
a=fmtp:97 octet-align=1
Received INVITE with SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.195
s=-
c=IN IP4 192.168.205.195
t=0 0
m=audio 13008 RTP/AVP 102 100 0 96 97 101 119
a=rtpmap:102 AMR-WB/16000/1
a=fmtp:102 mode-set=0,1,2
a=rtpmap:100 AMR/8000
a=fmtp:100 mode-set=0,2,5,7
a=rtpmap:0 PCMU/8000
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=fmtp:97 octet-align=1
a=rtpmap:101 telephone-event/16000
a=rtpmap:119 telephone-event/8000
My script is expecting fmtp: 101 0-15 but missing from received INVITE, When and in which case fmtp of DTMF should be expected and with what payload type of dynamic codec should we receive? What if I remove the fmtp expectation of telephone-event in the received INVITE from the script?
sip payload sdp
asked Mar 24 at 10:46
AnuAnu
459
We have a case where we have to apply the conditional codec policy on the egress side. But I have a problem with the script where sending telephone-event with payload-type 101 and 119 in the initial INVITE but did not received any fmtp for telephone-event whereas my scripts expects it to come. I am new to this field of SIP and SDP and unable to figure out the exact problem.
I thought the script is expecting what it shouldn't, so removed the expectation and the call is successfully completing. Below is the Sending and Received INVITE.
Sending INVITE with below SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.193
s=-
c=IN IP4 192.168.205.193
t=0 0
m=audio 10000 RTP/AVP 96 97 119
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:119 telephone-event/8000
a=fmtp:97 octet-align=1
Received INVITE with SDP:
v=0
o=user1 53655765 2353687637 IN IP4 192.168.205.195
s=-
c=IN IP4 192.168.205.195
t=0 0
m=audio 13008 RTP/AVP 102 100 0 96 97 101 119
a=rtpmap:102 AMR-WB/16000/1
a=fmtp:102 mode-set=0,1,2
a=rtpmap:100 AMR/8000
a=fmtp:100 mode-set=0,2,5,7
a=rtpmap:0 PCMU/8000
a=rtpmap:96 AMR/8000
a=rtpmap:97 AMR/8000
a=fmtp:97 octet-align=1
a=rtpmap:101 telephone-event/16000
a=rtpmap:119 telephone-event/8000
My script is expecting fmtp: 101 0-15 but missing from received INVITE, When and in which case fmtp of DTMF should be expected and with what payload type of dynamic codec should we receive? What if I remove the fmtp expectation of telephone-event in the received INVITE from the script?
sip payload sdp
sip payload sdp
asked Mar 24 at 10:46
AnuAnu
459
asked Mar 24 at 10:46
AnuAnu
459
asked Mar 24 at 10:46
AnuAnu
459
asked Mar 24 at 10:46
AnuAnu
459
asked Mar 24 at 10:46
AnuAnu
459
459
Please edit your question to make it clear: 1/ what side are you implementing? 2/ make a comparison between working scenario & non-working scenario? 3/ And finally, give an exact list of questions in your question...?
– AymericM
Mar 24 at 12:43
add a comment |
Please edit your question to make it clear: 1/ what side are you implementing? 2/ make a comparison between working scenario & non-working scenario? 3/ And finally, give an exact list of questions in your question...?
– AymericM
Mar 24 at 12:43
Please edit your question to make it clear: 1/ what side are you implementing? 2/ make a comparison between working scenario & non-working scenario? 3/ And finally, give an exact list of questions in your question...?
– AymericM
Mar 24 at 12:43
Please edit your question to make it clear: 1/ what side are you implementing? 2/ make a comparison between working scenario & non-working scenario? 3/ And finally, give an exact list of questions in your question...?
– AymericM
Mar 24 at 12:43
add a comment |
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Please edit your question to make it clear: 1/ what side are you implementing? 2/ make a comparison between working scenario & non-working scenario? 3/ And finally, give an exact list of questions in your question...?
– AymericM
Mar 24 at 12:43