Have SPA8000 connected to a Telco SIP gateway. Have a problem that Incoming Calls from IPHONES (started 4 weeks ago) ring but hang up on answer. other calls are OK from landlines and android.
(IP address have been replaced with names)
Trace from a IPHONE Call
call from IPHONE to SPA8000 (Fri 2021-11-12 05:56:44 UTC)
RingPartySipIP --> TelcoSIPSrv01 --> CalledPartySipIP, INVITE (SDP)
RingPartySipIP <-- TelcoSIPSrv01 <-- CalledPartySipIP, SIP: 100 Trying
RingPartySipIP <-- TelcoSIPSrv01 <-- CalledPartySipIP, SIP: 180 Ringing
.
TelcoSIPSrv01 <-- CalledPartySipIP, SIP: 200 OK (for INVITE) (SDP)
m=audio 16430 RTP/AVP 8 100 100
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:30
a=sendrecv
.
.
RingPartySipIP <-- TelcoSIPSrv01, SIP: 500 Server Internal Error
.
.
2021-11-12 05:56:46.944ISC TelcoSIPSrv01 The received SDP contained two rtpmap or fmtp lines for the same codec
.
Two rtpmap or fmtp lines have been found for the same codec.
The SDP failed to parse. This may cause call failure.
Payload ID: 100
Duplicated attribute: rtpmap
SDP:
m=audio 16430 RTP/AVP 8 100 100
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=ptime:30
a=sendrecv
Telco says this call failed because of Duplicated attribute: rtpmap
Customer's (SIP: 200 OK (for INVITE) (SDP)) has rtpmap:100 for two different things
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
This rtpmap:100 NSE/8000 is used for switching from voice to passthrough for fax/modem communication without involving SIP
and a=rtpmap:100 telephone-event/8000 <<<<< this is incorrect, for a telephone-event/8000 it should be a rtpmap:101, not 100
Now below is a good call sample:
View call from ANDROID to SPA8000 (Wed 2021-11-24 01:50:38 UTC)
RingPartySipIP --> TelcoSIPSrv01 --> CalledPartySipIP, INVITE (SDP)
RingPartySipIP <-- TelcoSIPSrv01 <-- CalledPartySipIP, SIP: 100 Trying
RingPartySipIP <-- TelcoSIPSrv01 <-- CalledPartySipIP, SIP: 180 Ringing
.
TelcoSIPSrv01 <-- CalledPartySipIP, SIP: 200 OK (for INVITE) (SDP)
m=audio 16410 RTP/AVP 8 100 101 <<<<<<<<<<<<
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000 <<<<<<<<<<<<
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
.
.
RingPartySipIP <-- TelcoSIPSrv01, SIP: 200 OK (for INVITE) (SDP)
m=audio 51938 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000 <<<<<<
a=fmtp:101 0-15
a=ptime:30
Does any one know what setting I need to change in the SPA800 to change 100 to 101 as the Telco is suggesting ???
