Hello, I'm using freeswitch and I'm experimenting with different hard- and softphones. I like the features of Linphone esp. regarding codec-support and easy way to also call sip-URL. All my softphones and hardware (Fritz-Box, Mitel OCX320 PBX as SIP-Client) can call each other inbound and outbound. With Linphone I only can do outbound calls, Codecs are ngotiated according to my preferences. No calls can be received.
This is the same for Linphone on Windows and Android. Here is a log (Linphone on Windows) which perhaps can show the problems. The call is not accepted (488) because the codecs don't match which is not understandable. IP's are anonymized, the switch is on Public IP and the phones connect from internal network (routed, no NAT involved). SIP/2.0 100 Trying Via: SIP/2.0/TLS 123.123.123.123:5081;rport;branch=z9hG4bKBp878UrQZD83Q From: "1010" <sip:[email protected]>;tag=j3ZK907Q034Ue To: <sip:[email protected]:60666;transport=tls> Call-ID: 4c4b9e43-d733-1234-c2ae-00505685f286 CSeq: 94948245 INVITE Content-Length: 0 message: 2016-08-07 13:16:37:085 New server dialog [05834550] , local tag [OzFMwZO], remote tag [j3ZK907Q034Ue] message: 2016-08-07 13:16:37:085 op [05D41280] : set_or_update_dialog() current=[00000000] new=[05834550] message: 2016-08-07 13:16:37:085 new incoming call from ["1010" <sip:[email protected]>] to [<sip:[email protected]:60666;transport=tls>] message: 2016-08-07 13:16:37:087 Found payload G722/8000 fmtp= message: 2016-08-07 13:16:37:087 Found payload PCMA/8000 fmtp= message: 2016-08-07 13:16:37:087 Found payload PCMU/8000 fmtp= message: 2016-08-07 13:16:37:087 Found payload telephone-event/8000 fmtp=0-16 error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AEAD_AES_256_GCM_8' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AEAD_AES_256_GCM_8' error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AEAD_AES_128_GCM_8' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AEAD_AES_128_GCM_8' error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AES_CM_256_HMAC_SHA1_80' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AES_CM_256_HMAC_SHA1_80' error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AES_CM_192_HMAC_SHA1_80' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AES_CM_192_HMAC_SHA1_80' message: 2016-08-07 13:16:37:087 Found valid crypto line (tag:5 algo:'AES_CM_128_HMAC_SHA1_80' key:'2x/OM3jI9/967s/9XuHss4BlRDDtTI/r+DDELEKl' error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AES_CM_256_HMAC_SHA1_32' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AES_CM_256_HMAC_SHA1_32' error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AES_CM_192_HMAC_SHA1_32' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AES_CM_192_HMAC_SHA1_32' message: 2016-08-07 13:16:37:087 Found valid crypto line (tag:8 algo:'AES_CM_128_HMAC_SHA1_32' key:'TJuCy3TKxyCDIwUjctk3FdJbfOhua6ZmFRBui2C3' error: 2016-08-07 13:16:37:087 Unsupported crypto suite 'AES_CM_128_NULL_AUTH' with parameters '' warning: 2016-08-07 13:16:37:087 Failed to parse crypto-algo: 'AES_CM_128_NULL_AUTH' message: 2016-08-07 13:16:37:087 Found: 2 valid crypto lines message: 2016-08-07 13:16:37:087 Found payload G722/8000 fmtp= message: 2016-08-07 13:16:37:087 Found payload PCMA/8000 fmtp= message: 2016-08-07 13:16:37:087 Found payload PCMU/8000 fmtp= message: 2016-08-07 13:16:37:087 Found payload telephone-event/8000 fmtp=0-16 message: 2016-08-07 13:16:37:088 Searching for already_a_call_with_remote_address. message: 2016-08-07 13:16:37:088 New LinphoneCall [05D6B488] initialized (LinphoneCore version: 3.9.1-1647-g25d2ac5) message: 2016-08-07 13:16:37:088 audio stream index found: 0, updating main audio stream index message: 2016-08-07 13:16:37:088 audio stream index found: 1, but main audio stream already set to 0 message: 2016-08-07 13:16:37:088 1 was used for video stream ; now using 3 message: 2016-08-07 13:16:37:088 check OS support for qwave.lib error: 2016-08-07 13:16:37:089 QOSAddSocketToFlow failed to add a flow with error 87 message: 2016-08-07 13:16:37:089 RtpSession bound to [0.0.0.0] ports [7078] [7079] message: 2016-08-07 13:16:37:089 rtp_session_enable_network_simulation:DISABLING NETWORK SIMULATION message: 2016-08-07 13:16:37:089 Configured srtp crypto suite: AES_CM_128_HMAC_SHA1_80 message: 2016-08-07 13:16:37:089 Configured srtp crypto suite: AES_CM_128_HMAC_SHA1_32 message: 2016-08-07 13:16:37:089 Configured srtp crypto suite: AES_256_CM_HMAC_SHA1_80 message: 2016-08-07 13:16:37:089 Configured srtp crypto suite: AES_256_CM_HMAC_SHA1_32 message: 2016-08-07 13:16:37:089 Creating ZRTP engine on rtp session [05897518] ssrc 0xe4cea744 message: 2016-08-07 13:16:37:090 Setting DSCP to 46 for MSAudio stream. message: 2016-08-07 13:16:37:090 Equalizer location: hp message: 2016-08-07 13:16:37:090 cannot set noise gate mode to [0] because no volume send message: 2016-08-07 13:16:37:090 Cannot determine multicast role for stream type [video] on call [05D6B488] message: 2016-08-07 13:16:37:090 check OS support for qwave.lib error: 2016-08-07 13:16:37:091 QOSAddSocketToFlow failed to add a flow with error 87 message: 2016-08-07 13:16:37:091 RtpSession bound to [0.0.0.0] ports [51346] [51347] message: 2016-08-07 13:16:37:091 rtp_session_enable_network_simulation:DISABLING NETWORK SIMULATION message: 2016-08-07 13:16:37:091 Initializing multistream ZRTP context on rtp session [05899D58] ssrc 0x8fd525b message: 2016-08-07 13:16:37:091 Setting DSCP to 0 for MSVideo stream. message: 2016-08-07 13:16:37:091 Cannot determine multicast role for stream type [text] on call [05D6B488] message: 2016-08-07 13:16:37:091 check OS support for qwave.lib error: 2016-08-07 13:16:37:091 QOSAddSocketToFlow failed to add a flow with error 87 message: 2016-08-07 13:16:37:091 RtpSession bound to [0.0.0.0] ports [11078] [11079] message: 2016-08-07 13:16:37:091 rtp_session_enable_network_simulation:DISABLING NETWORK SIMULATION message: 2016-08-07 13:16:37:131 Discovered mtu is 1500, RTP payload max size is 1440 message: 2016-08-07 13:16:37:131 Don't put video stream on local offer for call [05D6B488] message: 2016-08-07 13:16:37:131 Don't put text stream on local offer for call [05D6B488] message: 2016-08-07 13:16:37:131 Doing SDP offer/answer process of type incoming message: 2016-08-07 13:16:37:131 Declining mline 0, no corresponding stream in local capabilities description. message: 2016-08-07 13:16:37:131 No match for G722/8000/1 message: 2016-08-07 13:16:37:131 No match for PCMA/8000/1 message: 2016-08-07 13:16:37:131 No match for PCMU/8000/1 message: 2016-08-07 13:16:37:131 No match for telephone-event/8000/1 message: 2016-08-07 13:16:37:131 channel [05882BD0]: message sent to [TLS://mydomain.com:5081], size: [393] bytes SIP/2.0 488 Not acceptable here Via: SIP/2.0/TLS 123.123.123.123:5081;rport;branch=z9hG4bKBp878UrQZD83Q From: "1010" <sip:[email protected]>;tag=j3ZK907Q034Ue To: <sip:[email protected]:60666;transport=tls>;tag=OzFMwZO Call-ID: 4c4b9e43-d733-1234-c2ae-00505685f286 CSeq: 94948245 INVITE User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) Supported: replaces, outbound Content-Length: 0 -- *Ralf Schenk* fon +49 (0) 24 05 / 40 83 70 fax +49 (0) 24 05 / 40 83 759 mail *[email protected]* <mailto:[email protected]> *Databay AG* Jens-Otto-Krag-Straße 11 D-52146 Würselen *www.databay.de* <http://www.databay.de> Sitz/Amtsgericht Aachen • HRB:8437 • USt-IdNr.: DE 210844202 Vorstand: Ralf Schenk, Dipl.-Ing. Jens Conze, Aresch Yavari, Dipl.-Kfm. Philipp Hermanns Aufsichtsratsvorsitzender: Klaus Scholzen (RA) ------------------------------------------------------------------------
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