[asterisk-bugs] [JIRA] (ASTERISK-25924) PJSIP Polycom SRTP problem
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Fri Apr 15 09:55:57 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25924?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-25924:
------------------------------------
Assignee: Stacy Vinson
Status: Waiting for Feedback (was: Triage)
Please read through the linked Asterisk Issue Guidelines, remove the excessive debug from the description and attach it to the issue as .txt.
Please provide dialplan and SIP channel driver configuration necessary to reproduce the issue - as well as instructional steps for reproduction.
> 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
> Assignee: 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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