Hello!
I am using a Siemens HiPath8000 which is directly connected to my asterisk voicemailserver (via SIP).
When there is a incoming call from a standard HiPAth Extension (SIP) without enabled encryption everything is working perfectly:
######################
<-- SIP read from 172.28.11.2:5060:
INVITE sip:942@voippbx2.klu.com:5060;transport=UDP SIP/2.0
From: “USER” sip:123456789@172.28.11.2;user=phone;tag=snl-1213264126-6732141213937340-11
To: sip:942@voippbx2.com;user=phone
Via: SIP/2.0/UDP 172.28.11.2:5060;branch=z9hG4bKSNCLLC1213264126673286892693422
CSeq: 6758 INVITE
Contact: sip:123456789@172.28.11.2:5060;transport=udp;maddr=172.28.11.2
Call-ID: 6214623121-2281768495393121-11-592350184
X-Siemens-Call-type: ST-insecure
P-Asserted-Identity: “USER” sip:123456789@172.28.11.2
Max-Forwards: 69
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER
Call-Info: sip:123456789@172.28.11.2:0; purpose=CCS-Call
Accept-Language: en; q=0.0
Content-Type: application/sdp
Content-Length: 315
v=0
o=MxSIP 0 1617544244 IN IP4 172.28.13.3
s=SIP Call
c=IN IP4 172.28.13.3
t=0 0
m=audio 5012 RTP/AVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
— (15 headers 14 lines) —
Using INVITE request as basis request - 6214623121-2281768495393121-11-592350184
Sending to 172.28.11.2 : 5060 (non-NAT)
Found user '123456789’
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.28.13.3:5012
Found description format G722
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 942 in from-internal (domain voippbx2.klu.com)
list_route: hop: sip:123456789@172.28.11.2:5060;transport=udp;maddr=172.28.11.2
Transmitting (no NAT) to 172.28.11.2:5060:
SIP/2.0 100 Trying
#################
When there is a incoming call from a hipath8000 extension (SIP) with encryption enabled then I am getting a error message:
##################
<-- SIP read from 172.28.11.2:5060:
INVITE sip:942@voippbx2.klu.com:5060;transport=UDP SIP/2.0
From: “USER” sip:123456789@172.28.11.2;user=phone;tag=snl-1213264285-4543931213718678-11
To: sip:942@voippbx2.klu.com;user=phone
Via: SIP/2.0/UDP 172.28.11.2:5060;branch=z9hG4bKSNCLLC12132642854544671978508490
CSeq: 9387 INVITE
Contact: sip:123456789@172.28.11.2:5060;transport=udp;maddr=172.28.11.2
Call-ID: 5824623121-7423542357173121-11-592350184
X-Siemens-Call-type: ST-secure
P-Asserted-Identity: “USER” sip:123456789@172.28.11.2
Max-Forwards: 69
Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER
Call-Info: sip:123456789@172.28.11.2:0; purpose=CCS-Call
Accept-Language: en; q=0.0
Content-Type: application/sdp
Content-Length: 835
v=0
o=MxSIP 0 1600992348 IN IP4 172.28.13.3
s=SIP Call
c=IN IP4 172.28.13.3
t=0 0
a=key-mgmt:mikey AQAVgDjHmrACAAAAAAAAAAAAAABePjpxAAAAAAUBAAVtaWtleQsAy/twMgo9cAAKFM3/PMJS64kjQVM6+WptLw+vRRUNAQAAADYCAQEDBAAAAKAEBAAAAHALBAAAAFAAAQEBBAAAAIAJAQAGAQAFAQAIAQEKAQEHAQEMBAAAAAAAAAAkABAAEKWQ3yd6zlyWartNmNi+dNUADh5JWj8l9v1oRWRA2i/LAA==
m=audio 5012 RTP/AVP 9 8 0 18 9 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
m=audio 5010 RTP/SAVP 9 8 0 18 9 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
— (15 headers 26 lines) —
Using INVITE request as basis request - 5824623121-7423542357173121-11-592350184
Sending to 172.28.11.2 : 5060 (non-NAT)
Found user '123456789’
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Transmitting (no NAT) to 172.28.11.2:5060:
SIP/2.0 488 Not acceptable here
###########################
As far as I understand the SDP is now extended with the options of SRTP. But Asterisk should take the options for RTP, correct?
Best Regards,
Johannes