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From: | Ghislain MARY |
Subject: | Re: [Linphone-developers] Linphone not replying to SDP packets |
Date: | Tue, 25 Mar 2014 10:56:07 +0100 |
User-agent: | Mozilla/5.0 (X11; Linux x86_64; rv:24.0) Gecko/20100101 Icedove/24.3.0 |
Hi,
The retransmissions of 401 are simply UDP retransmissions. The error you are getting comes from the 200 OK messages your Asterisk server is sending that do not contain the mandatory contact header. For example, a bad 200 OK in your logs: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.239.216.149:39383;branch=z9hG4bK.GIDiMHiyC;received=199.123.195.94;rport=44213 From: <sip:address@hidden>;tag=CbeXBXXD~ To: sip:address@hidden;tag=as61386cd3 Call-ID: sxQFf3e5Lm CSeq: 21 INVITE Server: VoipCoVoIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 273 v=0 o=root 799165012 799165012 IN IP4 132.39.8.242 s=VoipCoVoIP c=IN IP4 132.39.8.242 t=0 0 m=audio 6140 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecvAnd a correct one: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.239.216.149:39383;branch=z9hG4bK.qshxnPn1j;received=199.123.195.94;rport=44213 From: <sip:address@hidden>;tag=y47PigZUM To: sip:address@hidden;tag=as3148b844 Call-ID: N7rFteUp~U CSeq: 21 INVITE Server: VoipCoVoIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:address@hidden:5060> Content-Type: application/sdp Content-Length: 276 v=0 o=root 1145458109 1145458109 IN IP4 132.39.8.242 s=VoipCoVoIP c=IN IP4 132.39.8.242 t=0 0 m=audio 28266 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecvCheers, Ghislain On 24/03/2014 23:21, list traffic wrote:
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