SRTP issue on some hardware phones

Asterisk Asterisk SIP
1

Hello,
I have issue with SRTP. When using softphone connected to my asterisk server, SRTP works ok.
If i try calling from a Grandstream GXP2130 phone with enabled SRTP my server shows 488 Not Acceptable Here.
On my GXP2130 if i disable SRTP it works fine.

Below are sample sip messages

  1. GXP2130, SRTP enabled, 100 Trying and then failing with 488 Not acceptable here:

INVITE sip:066*******@voip*****:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.103:52665;branch=z9hG4bK1718427006;rport;alias
From: sip:103@voip:5061******;tag=1483956476
To: sip:066*******@voip:5061*****
Call-ID: 502091212-5063-4@BJC.BGI.BAA.BAD
CSeq: 21 INVITE
Contact: sips:103@192.168.100.103:52665;transport=tls
Authorization: Digest username=“103”, realm=“asterisk”, nonce=“1573049009/ead44f581fdcfa1ba3d82502a06da4be”, uri="sip:066******$…
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.9.26
Privacy: none
P-Preferred-Identity: sip:103@voip:5061*******
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 601

v=0
o=103 8000 8000 IN IP4 192.168.100.103
s=SIP Call
c=IN IP4 192.168.100.103
t=0 0
m=audio 5204 RTP/SAVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_256_HMAC_SHA1_80 inline:F/0scmDP6keRNoDEMPMLwWOoYPztu/S+wfWZOcsPsOIM3VRsrD+0PnU0AqYnDWeL
a=crypto:2 AES_CM_256_HMAC_SHA1_32 inline:tciHooN7YERx+n48CS4fFQtzgbiyNfYoavjOkQU1HLr9pFyAH73FkLdDzcBx7NV9
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:thrc7R2DH9NScK45FJf+Vg4oqLtei6wKC3HUmv9B
a=crypto:4 AES_CM_128_HMAC_SHA1_32 inline:h7VbY6N458JMOTL6ckeRcJ2fmUVa99AGAdt41nV3

  1. Same GXP2130, SRTP disabled, works OK:

INVITE sip:066***********@voip*****:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.100.103:42812;branch=z9hG4bK1533293156;rport;alias
From: sip:103@voip:5061*****;tag=1693902311
To: sip:066******@voip:5061*****
Call-ID: 2016177651-5063-4@BJC.BGI.BAA.BAD
CSeq: 21 INVITE
Contact: sips:103@192.168.100.103:42812;transport=tls
Authorization: Digest username=“103”, realm=“asterisk”, nonce=“1573047675/54e46990918ec4164665e46c128bf107”, uri="sip:066*****$…
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.9.26
Privacy: none
P-Preferred-Identity: sip:103@voip:5061******
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 216

v=0
o=103 8000 8000 IN IP4 192.168.100.103
s=SIP Call
c=IN IP4 192.168.100.103
t=0 0
m=audio 5204 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

  1. Dial from softphone, SRTP enabled, works OK:

INVITE sip:098********@voip*****;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.0.39:51240;branch=z9hG4bK-524287-1—34c6db2bf82045ca;rport
Max-Forwards: 70
Contact: sip:105@77:46919******;transport=TLS
To: sip:098*******@voip*****
From: sip:105@voip*****;transport=TLS;tag=06732823
Call-ID: IIWglrKr4bqFwDsZao8Iew…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.3.0
Authorization: Digest username=“105”,realm=“asterisk”,nonce=“1573052113/773eab29d5c2a7f0defb5b317ec7085b”,uri="sip:098*****$
Allow-Events: presence, kpml, talk
Content-Length: 1243

v=0
o=Zoiper 1573052112839 1 IN IP4 192.168.0.39
s=Z
c=IN IP4 192.168.0.39
t=0 0
m=audio 36212 RTP/SAVP 8 101 0 3 97 110
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:110 speex/8000
a=sendrecv
a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHRC+uZnNIAR6g==
a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHRC+uZnNIAR6g==
a=crypto:9 AES_CM_256_HMAC_SHA1_80 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHRC+uZnNIAR6g==
a=crypto:10 AES_CM_256_HMAC_SHA1_32 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHRC+uZnNIAR6g==
a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHQ=
a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHQ=
a=crypto:7 AES_CM_192_HMAC_SHA1_80 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHQ=
a=crypto:8 AES_CM_192_HMAC_SHA1_32 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas41UCEu9H0NHQ=
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas4
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Vmb208g9gmTVMI+VMDTqVgDCvOKPhkzGQBVwWas4

Any suggestions why my asterisk server is not working with GXP2130 SRTP enabled?
How can i debug “488 Not acceptable here” message to see what exactly is not acceptable?

2

SRTP has to be explicitly enabled in Asterisk. What channel driver are you using? What is the configuration for the SIP user/peer or endpoint? Is SRTP enabled for it?

3

jcolp, i was about to post the config for the [103] user and seen it has media_encryption=no
Thank you very much.

4

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