[asterisk-bugs] [JIRA] (ASTERISK-25924) PJSIP Polycom SRTP problem
Stacy Vinson (JIRA)
noreply at issues.asterisk.org
Thu Apr 14 13:15:56 CDT 2016
Stacy Vinson created ASTERISK-25924:
---------------------------------------
Summary: PJSIP Polycom SRTP problem
Key: ASTERISK-25924
URL: https://issues.asterisk.org/jira/browse/ASTERISK-25924
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Channels/chan_pjsip
Affects Versions: 13.8.0
Environment: ubuntu server 14.04 x64
Polycom VVX 600
Reporter: Stacy Vinson
Severity: Minor
I'm having a problem getting my Polycom VVX 600 working with SRTP and PJSIP I get a 488 after the RTP/SAVP.
The Polycom works fine with SRTP if i use chan_sip, and my AASTRA 57i works fine with PJSIP and SRTP.
The Polycom also works fine with PJSIP if i disable SRTP.
I also tested a few soft phones with SRTP enabled on PJSIP and did not have any problems. so it looks like it's only a
problem with the polycom phones.
Below is PJSIP debug:
<--- Received SIP request (615 bytes) from TLS:77.23.194.200:65369 --->
ACK sip:username at asterisk.domain.com:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.105.122:65369;branch=z9hG4bKc53e2bcbDC59CA7C
From: "username" <sip:username at asterisk.domain.com>;tag=F419090D-6D823456
To: <sip:username at asterisk.domain.com;user=phone>;tag=z9hG4bKc53e2bcbDC59CA7C
CSeq: 1 ACK
Call-ID: 3485c39-ff8bf692-bbb2bdd7 at 192.168.105.122
Contact: <sip:username at 192.168.105.122:65369;transport=tls>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: Polycom/5.4.2.2834 PolycomVVX-VVX_600-UA/5.4.2.2834
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
<--- Received SIP request (3206 bytes) from TLS:77.23.194.200:65369 --->
INVITE sip:username at asterisk.domain.com:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.105.122:65369;branch=z9hG4bK45fd2618CADE0025
From: "username" <sip:username at asterisk.domain.com:5061>;tag=F419090D-6D823456
To: <sip:username at asterisk.domain.com;user=phone>
CSeq: 2 INVITE
Call-ID: 3485c39-ff8bf692-bbb2bdd7 at 192.168.105.122
Contact: <sip:username at 192.168.105.122:65369;transport=tls>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: Polycom/5.4.2.2834 PolycomVVX-VVX_600-UA/5.4.2.2834
Accept-Language: en
Supported: replaces,100rel
Allow-Events: conference,talk,hold
Authorization: Digest username="username", realm="asterisk", nonce="1460417306/23d0bb07ec54db4679896d36b78ad889", qop=auth, cnonce="6cpUdjTeq4XtB6D", nc=00000001, opaque="42b41f285ec9c27b", uri="sip:username at asterisk.domain.com:5061;user=phone;transport=tls", response="1a85022c8a3299f6cbe17d959e08956f", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 2203
v=0
o=- 1460417315 1460417315 IN IP4 192.168.105.122
s=Polycom IP Phone
c=IN IP4 77.23.194.200
t=0 0
a=sendrecv
m=audio 2430 RTP/SAVP 9 115 8 0 102 18 127
a=crypto:15 AES_CM_128_HMAC_SHA1_80 inline:QUreV3PLt2teayweBKs1WmVaOvgerH4qTNG+E7z+|2^31
a=rtpmap:9 G722/8000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
a=ice-pwd:vxfcilNqkXpzK7hCJ0vgM5zo/Eyeball
a=ice-ufrag:AF5GjP
a=rtcp:2431
a=candidate:1 1 UDP 2130706431 192.168.105.122 2230 typ host
a=candidate:1 2 UDP 2130706430 192.168.105.122 2231 typ host
a=candidate:2 1 TCP 2120810239 192.168.105.122 48324 typ host tcptype passive
a=candidate:2 2 TCP 2120810238 192.168.105.122 59055 typ host tcptype passive
a=candidate:3 1 TCP 2121072639 192.168.105.122 48324 typ host tcptype active
a=candidate:3 2 TCP 2121072638 192.168.105.122 59055 typ host tcptype active
a=candidate:4 1 UDP 1694498815 77.23.194.200 2430 typ srflx raddr 192.168.105.122 rport 2430
a=candidate:4 2 UDP 1694498814 77.23.194.200 2431 typ srflx raddr 192.168.105.122 rport 2431
m=audio 2430 RTP/AVP 9 115 8 0 102 18 127
a=rtpmap:9 G722/8000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
a=ice-pwd:vxfcilNqkXpzK7hCJ0vgM5zo/Eyeball
a=ice-ufrag:AF5GjP
a=rtcp:2431
a=candidate:1 1 UDP 2130706431 192.168.105.122 2230 typ host
a=candidate:1 2 UDP 2130706430 192.168.105.122 2231 typ host
a=candidate:2 1 TCP 2120810239 192.168.105.122 48324 typ host tcptype passive
a=candidate:2 2 TCP 2120810238 192.168.105.122 59055 typ host tcptype passive
a=candidate:3 1 TCP 2121072639 192.168.105.122 48324 typ host tcptype active
a=candidate:3 2 TCP 2121072638 192.168.105.122 59055 typ host tcptype active
a=candidate:4 1 UDP 1694498815 77.23.194.200 2430 typ srflx raddr 192.168.105.122 rport 2430
a=candidate:4 2 UDP 1694498814 77.23.194.200 2431 typ srflx raddr 192.168.105.122 rport 2431
<--- Transmitting SIP response (396 bytes) to TLS:77.23.194.200:65369 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 192.168.105.122:65369;rport=65369;received=77.23.194.200;branch=z9hG4bK45fd2618CADE0025
Call-ID: 3485c39-ff8bf692-bbb2bdd7 at 192.168.105.122
From: "username" <sip:username at asterisk.domain.com>;tag=F419090D-6D823456
To: <sip:username at asterisk.domain.com;user=phone>;tag=30214dc9-3050-42fc-8ac5-2e1ab7b5edd4
CSeq: 2 INVITE
Server: Asterisk PBX
Content-Length: 0
<--- Received SIP request (628 bytes) from TLS:77.23.194.200:65369 --->
ACK sip:username at asterisk.domain.com:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.105.122:65369;branch=z9hG4bK45fd2618CADE0025
From: "username" <sip:username at asterisk.domain.com>;tag=F419090D-6D823456
To: <sip:username at asterisk.domain.com;user=phone>;tag=30214dc9-3050-42fc-8ac5-2e1ab7b5edd4
CSeq: 2 ACK
Call-ID: 3485c39-ff8bf692-bbb2bdd7 at 192.168.105.122
Contact: <sip:username at 192.168.105.122:65369;transport=tls>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: Polycom/5.4.2.2834 PolycomVVX-VVX_600-UA/5.4.2.2834
Accept-Language: en
Max-Forwards: 70
Content-Length: 0
******************************************************************************
The polycom works fine on chan_sip:
chan_sip debug:
<--- SIP read from TLS:77.23.194.200:48093 --->
INVITE sip:username at 92.21.193.10:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.105.122:48093;branch=z9hG4bKfafbe35482299131
From: "username" <sip:username at asterisk.domain.com:5061>;tag=7A82C553-7F04AC64
To: <sip:username at asterisk.domain.com;user=phone>;tag=as77212be2
CSeq: 4 INVITE
Call-ID: 9c11e3df-a8418a80-84ec976d at 192.168.105.122
Contact: <sip:username at 192.168.105.122:48093;transport=tls>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: Polycom/5.4.2.2834 PolycomVVX-VVX_600-UA/5.4.2.2834
Accept-Language: en
Supported: replaces,100rel
Allow-Events: conference,talk,hold
Authorization: Digest username="username", realm="asterisk", nonce="4c14a874", uri="sip:username at asterisk.domain.com:5061;user=phone;transport=tls", response="f4773acfc75ebdd81dc86fb9b9100516", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 1441
v=0
o=- 1460417442 1460417444 IN IP4 192.168.105.122
s=Polycom IP Phone
c=IN IP4 77.23.194.200
t=0 0
m=audio 2232 RTP/SAVP 9 115 8 0 102 18 127
a=crypto:16 AES_CM_128_HMAC_SHA1_80 inline:xG4V5WzVttt2RNQFzKxST+ahxiD4MeMfsEmY2Blm|2^31
a=rtpmap:9 G722/8000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
a=ice-pwd:t5wb1HuskjjjHNdCfumyc640/Eyeball
a=ice-ufrag:AFoAd3
a=rtcp:2233
a=remote-candidates:1 92.21.193.10 11908 2 92.21.193.10 11909
a=candidate:1 1 UDP 1862270975 77.23.194.200 2232 typ prflx raddr 192.168.105.122 rport 2232
a=candidate:1 2 UDP 1862270974 77.23.194.200 2233 typ prflx raddr 192.168.105.122 rport 2233
m=audio 2232 RTP/AVP 9 115 8 0 102 18 127
a=rtpmap:9 G722/8000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
a=ice-pwd:t5wb1HuskjjjHNdCfumyc640/Eyeball
a=ice-ufrag:AFoAd3
a=rtcp:2233
a=remote-candidates:1 92.21.193.10 11908 2 92.21.193.10 11909
a=candidate:1 1 UDP 1862270975 77.23.194.200 2232 typ prflx raddr 192.168.105.122 rport 2232
a=candidate:1 2 UDP 1862270974 77.23.194.200 2233 typ prflx raddr 192.168.105.122 rport 2233
<------------->
--- (16 headers 40 lines) ---
Sending to 77.23.194.200:48093 (NAT)
Found RTP audio format 9
Found RTP audio format 115
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 102
Found RTP audio format 18
Found RTP audio format 127
Found audio description format G722 for ID 9
Found audio description format G7221 for ID 115
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G7221 for ID 102
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 127
[Apr 11 18:30:36] WARNING[23993][C-00000000]: chan_sip.c:10270 process_sdp: Declining non-primary audio stream: audio 2232 RTP/AVP 9 115 8 0 102 18 127
Capabilities: us - (g722|ulaw|alaw), peer - audio=(ulaw|alaw|g722|g729|siren7|siren14)/video=(nothing)/text=(nothing), combined - (g722|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 77.23.194.200:2232
<--- Transmitting (NAT) to 77.23.194.200:48093 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.105.122:48093;branch=z9hG4bKfafbe35482299131;received=77.23.194.200;rport=48093
From: "username" <sip:username at asterisk.domain.com:5061>;tag=7A82C553-7F04AC64
To: <sip:username at asterisk.domain.com;user=phone>;tag=as77212be2
Call-ID: 9c11e3df-a8418a80-84ec976d at 192.168.105.122
CSeq: 4 INVITE
Server: Asterisk PBX 13.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:username at 92.21.193.10:5061;transport=TLS>
Content-Length: 0
<------------>
Audio is at 11908
Adding codec g722 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pjsip endpoint config:
type=endpoint
media_encryption=sdes
srtp_tag_32=no
tos_audio=ef
tos_video=af41
context=inside
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
ice_support=yes
mailboxes=(REMOVED)@default
mwi_from_user=(REMOVED)
callerid=(REMOVED)
;device_state_busy_at=2
allow_subscribe=yes
;sub_min_expiry=30
call_group=3
pickup_group=1-3
disallow=all
allow=g722
allow=ulaw
allow=alaw
aors=(REMOVED)
auth=(REMOVED)
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