Hi, Bogdan. Here is my simple scenario description:
UA1----Opensips----Asterisk ---- Opensips ----UA2 Transport protocol doesn't change during this chain and it's tls, if I understand you right. I attached SIP capture of the call. As you can see, there is the dynamic tcp port in the RR hrd of last reply to client from which Opensips connected to the Asterisk. Instead of one, to which UA1 connected to Opensips (5061). As a result, there is a media session between UAs, but only for 30 sec, during of which the UA1 tried to send ACK to the Opensips, but unsuccessfully for quite clear reason. Is there the resolution how to realize this scenario without rewriting RR? Best Regards, Sergey Pysanko. [image: Mailtrack] <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;> Sender notified by Mailtrack <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;> 01/04/22, 01:46:49 PM вт, 4 янв. 2022 г. в 11:47, Bogdan-Andrei Iancu <[email protected]>: > Hi Sergey, > > Manually altering the RR hdr is a receipt for disaster :). Somehow I > suspect you do not do double RR (as the protocol changes for the call). > This double RR is automatically done (by default) when doing > `record_route()`. Do you get 2 RR hdrs when routing the initial INVITE ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 2021 > https://opensips.org/training/OpenSIPS_eBootcamp_2021/ > > On 1/4/22 11:27 AM, Sergey Pisanko wrote: > > Hello, Bogdan, . > > Thank you for your answer. I've solved my issue recently just rewriting > Record - Route header with appropriate port within "onreply route block" by > subst function. > > Best Regards, > Sergey Pysanko. > > > > [image: Mailtrack] > <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;> > Sender > notified by > Mailtrack > <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;> > 01/04/22, > 11:27:07 AM > > пн, 3 янв. 2022 г. в 17:59, Bogdan-Andrei Iancu <[email protected]>: > >> Hello Sergey, >> >> Could you provide a SIP capture (and calling scenario) to underline the >> issue you have ? >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 2021 >> https://opensips.org/training/OpenSIPS_eBootcamp_2021/ >> >> On 12/30/21 2:50 PM, Sergey Pisanko wrote: >> >> Hello! >> >> I try to realize the next scenario with UAs, Opensips-2.4 and Asterisk. >> UAs are registered onto Asterisk through Opensips and also - on Opensips >> if the 200 OK is came back from Asterisk. >> Calls between UAs are relayed to Asterisk by Opensips. >> This scenario works fine with udp. But it needs to do with tls. And here >> I have the problem. What happens. >> Unlike udp, tcp cannot listen its port and create clients connection at >> the same time. Opensips listens tls port for clients connection >> whereas it creates dynamic tcp port to connect to Asterisk. As a result, >> I see that port in Record-Route header in 200 OK addressed to caller. >> Thus, callers ACK comes to that dynamic port instead of Opensips listened >> port and Opensips dropped it. >> And question is how to force Opensips to put right port for caller? >> >> Regards, >> Serhii Pysanko. >> >> >> >> [image: Mailtrack] >> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;> >> Sender >> notified by >> Mailtrack >> <https://mailtrack.io?utm_source=gmail&utm_medium=signature&utm_campaign=signaturevirality11&;> >> 12/30/21, >> 02:49:47 PM >> >> _______________________________________________ >> Users mailing >> [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
---------------------------------- UA1 to OPENSIPS Opensips_IP:5061: ---------------------------------- INVITE sip:508080@asterisk_IP:5062;transport=tls SIP/2.0 Via: SIP/2.0/TLS UA1_IP:52732;rport;branch=z9hG4bKPjb0cc2bffdc81454eb26583c7d3925484;alias Max-Forwards: 70 From: <sip:505050@asterisk_IP>;tag=d8e0d49a268d4b51aa85b8f79d2dc062 To: <sip:508080@asterisk_IP> Contact: <sip:505050@UA1_IP:52732;transport=TLS;ob> Call-ID: adb91a11aeb84cebbb1a0e9b3f737f9c CSeq: 9147 INVITE Route: <sip:Opensips_IP:5061;transport=tls;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.20.5 Authorization: Digest username="505050", realm="asterisk", nonce="1641298342/d7e3fb8b3d36162e02c6f06ba7ce025d", uri="sip:508080@asterisk_IP:5062;transport=tls", response="ef89c414b398b86f836c647a4618716e", algorithm=md5, cnonce="882c1c44e3634a188fee3c3cfcea4893", opaque="6a1b7faf68185146", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 693 v=0 o=- 3850294342 3850294342 IN IP4 UA1_IP s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4000 RTP/SAVP 8 0 96 101 102 c=IN IP4 UA1_IP b=TIAS:64000 a=rtcp:4001 IN IP4 UA1_IP a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 opus/48000/2 a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000;useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:102 telephone-event/48000 a=fmtp:102 0-16 a=ssrc:427376828 cname:51ff0c590e976fab a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sk7/Eszxo1FzEK7yX2uLJ31YtX41abwzMzB9+TLI a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:pYLtDmaE+aUjy1+3orAeDjO9nonOJo7M2DuTSO6k ------------------------------- Opensips (48470) to Asterisk (5062) ------------------------------- <--- Received SIP request (2110 bytes) from TLS:Opensips_IP:48470 ---> INVITE sip:508080@asterisk_IP:5062;transport=tls SIP/2.0 Record-Route: <sip:Opensips_IP:5061;transport=tls;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062> Via: SIP/2.0/TLS Opensips_IP:5061;branch=z9hG4bKb8a2.9fe1b065.0;i=9b32ca86 Via: SIP/2.0/TLS UA1_IP:52732;received=UA1_IP;rport=52732;branch=z9hG4bKPjb0cc2bffdc81454eb26583c7d3925484;alias Max-Forwards: 68 From: <sip:505050@asterisk_IP>;tag=d8e0d49a268d4b51aa85b8f79d2dc062 To: <sip:508080@asterisk_IP> Contact: <sip:505050@UA1_IP:52732;transport=TLS;ob> Call-ID: adb91a11aeb84cebbb1a0e9b3f737f9c CSeq: 9147 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.20.5 Authorization: Digest username="505050", realm="asterisk", nonce="1641298342/d7e3fb8b3d36162e02c6f06ba7ce025d", uri="sip:508080@asterisk_IP:5062;transport=tls", response="ef89c414b398b86f836c647a4618716e", algorithm=md5, cnonce="882c1c44e3634a188fee3c3cfcea4893", opaque="6a1b7faf68185146", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 899 v=0 o=- 3850294342 3850294342 IN IP4 UA1_IP s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 24120 RTP/SAVP 8 0 96 101 102 c=IN IP4 Opensips_IP b=TIAS:64000 a=ssrc:427376828 cname:51ff0c590e976fab a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=rtpmap:102 telephone-event/48000 a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000;useinbandfec=1 a=fmtp:101 0-16 a=fmtp:102 0-16 a=sendrecv a=rtcp:24121 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sk7/Eszxo1FzEK7yX2uLJ31YtX41abwzMzB9+TLI a=setup:actpass a=fingerprint:sha-1 E8:4F:19:70:C7:74:B5:73:7A:65:2C:60:1C:0F:72:B6:C9:D6:44:B7 a=ice-ufrag:ORyCsYFJ a=ice-pwd:YWCdb2Dnreyoz8P0IU54yPAR8r a=candidate:IEU3ysrf067x3ln1 1 UDP 2130706431 Opensips_IP 24120 typ host a=candidate:IEU3ysrf067x3ln1 2 UDP 2130706430 Opensips_IP 24121 typ host --------------------------- Asterisk to Opensips reply --------------------------- <--- Transmitting SIP response (1465 bytes) to TLS:Opensips_IP:48470 ---> SIP/2.0 200 OK Via: SIP/2.0/TLS Opensips_IP:5061;rport=48470;received=Opensips_IP;branch=z9hG4bKb8a2.9fe1b065.0;i=9b32ca86 Via: SIP/2.0/TLS UA1_IP:52732;rport=52732;received=UA1_IP;branch=z9hG4bKPjb0cc2bffdc81454eb26583c7d3925484;alias Record-Route: <sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062> Call-ID: adb91a11aeb84cebbb1a0e9b3f737f9c From: <sip:505050@Asterisk_IP>;tag=d8e0d49a268d4b51aa85b8f79d2dc062 To: <sip:508080@Asterisk_IP>;tag=3d503b27-cbd7-49e9-b076-49686b709c95 CSeq: 9147 INVITE Server: FPBX-14.0.16.11(16.19.0) Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Contact: <sip:Asterisk_IP:5062;transport=TLS> Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer P-Asserted-Identity: "508080" <sip:508080@Asterisk_IP> Content-Type: application/sdp Content-Length: 477 v=0 o=- 3850294342 3850294344 IN IP4 Asterisk_IP s=Asterisk c=IN IP4 Asterisk_IP t=0 0 m=audio 30410 RTP/SAVP 96 0 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:umKjN3UNSUtW+Cn/R2uStQZBhIWgJeDVyO2T5gCs a=rtpmap:96 opus/48000/2 a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000;useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:20 a=sendrecv --------------------------- Opensips to UA1 reply --------------------------- SIP/2.0 200 OK Via: SIP/2.0/TLS UA1_IP:52732;rport=52732;received=UA1_IP;branch=z9hG4bKPjb0cc2bffdc81454eb26583c7d3925484;alias Record-Route: <sip:Opensips_IP:48470;transport=TLS;lr;ftag=d8e0d49a268d4b51aa85b8f79d2dc062> Call-ID: adb91a11aeb84cebbb1a0e9b3f737f9c From: <sip:505050@Asterisk_IP>;tag=d8e0d49a268d4b51aa85b8f79d2dc062 To: <sip:508080@Asterisk_IP>;tag=3d503b27-cbd7-49e9-b076-49686b709c95 CSeq: 9147 INVITE Server: FPBX-14.0.16.11(16.19.0) Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Contact: <sip:Asterisk_IP:5062;transport=TLS> Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer P-Asserted-Identity: "508080" <sip:508080@Asterisk_IP> Content-Type: application/sdp Content-Length: 639 v=0 o=- 3850294342 3850294344 IN IP4 Asterisk_IP s=Asterisk c=IN IP4 Opensips_IP t=0 0 m=audio 24138 RTP/SAVP 96 0 8 101 a=maxptime:20 a=rtpmap:96 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:96 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=64000;useinbandfec=1 a=fmtp:101 0-16 a=sendrecv a=rtcp:24139 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:umKjN3UNSUtW+Cn/R2uStQZBhIWgJeDVyO2T5gCs a=ptime:20 a=candidate:IEU3ysrf067x3ln1 1 UDP 2130706431 Opensips_IP 24138 typ host a=candidate:IEU3ysrf067x3ln1 2 UDP 2130706430 Opensips_IP 24139 typ host
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