[asterisk-bugs] [JIRA] (ASTERISK-25924) PJSIP Polycom SRTP problem
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Thu Apr 14 13:15:56 CDT 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25924?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230264#comment-230264 ]
Asterisk Team commented on ASTERISK-25924:
------------------------------------------
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.
A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
> 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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