Re: [OpenSIPS-Users] double sdp.
Jeff, I think that I should call only once rtpengine_offer (then the sdp is allright). But I forgot to mention that the port that the A side sees should not change meaning that whatever I respond on the received 183 should remain the same on the 200 OK. Therefore I wonder, if the fix wouldn't be to call only rtpengine_answer a second time upon receiving the 200 OK from C. Anyway, I will test that tomorrow morning and come back on this. wkr, Outlook voor iOS <https://aka.ms/o0ukef> downloaden -- *Van:* Users namens Jeff Pyle < j...@ugnd.org> *Verzonden:* Monday, October 18, 2021 11:50:22 PM *Aan:* OpenSIPS users mailling list *Onderwerp:* Re: [OpenSIPS-Users] double sdp. Johan, To avoid problems like this, I call rtpengine_offer() in branch_route on initial invites, and make sure to call rtpengine_delete() in any failure route to remove any session from a failed offer that was never used. Perhaps these will help in your situation as well. - Jeff On Mon, Oct 18, 2021 at 11:37 AM Johan De Clercq wrote: Hi, A and B are on the same proxy. A calls B, (as I need transcoding I need to call rtpengine_offer here) B returns 183 with SDP. (this implies calling rtpengine_answer in onreply_route) B lets the call time out On the proxy I intercept the 480 returned by B and I change the INVITE so that it point to SEMS (this ismplies calling rtpengine_offer again) Issue: when you call 2x rtpengine_offer, you end up with a double sdp body. So, how can I instruct opensips to overwrite the body instead of appending one ? wkr, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] double sdp.
Johan, To avoid problems like this, I call rtpengine_offer() in branch_route on initial invites, and make sure to call rtpengine_delete() in any failure route to remove any session from a failed offer that was never used. Perhaps these will help in your situation as well. - Jeff On Mon, Oct 18, 2021 at 11:37 AM Johan De Clercq wrote: > Hi, > > A and B are on the same proxy. > > A calls B, > (as I need transcoding I need to call rtpengine_offer here) > B returns 183 with SDP. > (this implies calling rtpengine_answer in onreply_route) > B lets the call time out > On the proxy I intercept the 480 returned by B > and I change the INVITE so that it point to SEMS > (this ismplies calling rtpengine_offer again) > > Issue: when you call 2x rtpengine_offer, you end up with a double sdp > body. > > So, > how can I instruct opensips to overwrite the body instead of appending one > ? > > wkr, > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] double sdp.
Hi, A and B are on the same proxy. A calls B, (as I need transcoding I need to call rtpengine_offer here) B returns 183 with SDP. (this implies calling rtpengine_answer in onreply_route) B lets the call time out On the proxy I intercept the 480 returned by B and I change the INVITE so that it point to SEMS (this ismplies calling rtpengine_offer again) Issue: when you call 2x rtpengine_offer, you end up with a double sdp body. So, how can I instruct opensips to overwrite the body instead of appending one ? wkr, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double SDP
Thanks! Now all is OK! На ср, 4.09.2019 г. в 15:34 ч. Ben Newlin написа: > If you don't want to have both in the second INVITE, you can try putting > both rtpengine_offer calls in branch routes instead. I haven't worked with > rtpengine, but with other messages changes like this if you place them in > the branch route then they affect only the current branch; after failure > the original message will be returned and you may then be able to add > RTP/SAVP only. > > Ben Newlin > > On 9/4/19, 8:27 AM, "Users on behalf of Alexey Vasilyev" < > users-boun...@lists.opensips.org on behalf of alexei.vasil...@gmail.com> > wrote: > > This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. > This is > Invite from snom phone, for example: > > Sent to tls:135.42.212.82:5061 at Sep 4 14:19:18.641 (1383 bytes): > > INVITE sip:*7...@sip.test.dk SIP/2.0 > Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport > From: "Demo" ;tag=ncsplp1nvz > To: > Call-ID: 313536373539393535363232353137-eewp9wlm45rf > CSeq: 2 INVITE > Max-Forwards: 70 > User-Agent: snom320/8.7.5.44 > Contact: ;reg-id=1 > X-Serialnumber: 000XXX > P-Key-Flags: keys="3" > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, > MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 600 > Min-SE: 90 > Authorization: Digest > username="200",realm="asterisk",nonce="7b2d56ec",uri="sip:* > 7...@sip.test.dk",response="7a9fe1f24a6f7585fb7323237a000167",algorithm=MD5 > Content-Type: application/sdp > Content-Length: 476 > > v=0 > o=root 558099897 558099897 IN IP4 172.16.1.29 > s=call > c=IN IP4 172.16.1.29 > t=0 0 > m=audio 60812 RTP/SAVP 9 8 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ysn2nTlXXAuZYcpOhf1g/h+oG > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > m=audio 60812 RTP/AVP 9 8 101 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > > > > > - > --- > Alexey Vasilyev > -- > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double SDP
Hi Dragomir. May be you miss rtpengine_answer on reply?
Like
onreply_route[handle_reply] {
if (has_body("application/sdp"){
$var(rtpengine_flags) = "RTP/AVP replace-session-connection
replace-origin ICE=remove";
}
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Re: [OpenSIPS-Users] Double SDP
Hello, Thanks for your replay. If that's OK. Why do I have one-way voice? Rerards, На ср, 4.09.2019 г. в 15:29 ч. Alexey Vasilyev написа: > This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This > is > Invite from snom phone, for example: > > Sent to tls:135.42.212.82:5061 at Sep 4 14:19:18.641 (1383 bytes): > > INVITE sip:*7...@sip.test.dk SIP/2.0 > Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport > From: "Demo" ;tag=ncsplp1nvz > To: > Call-ID: 313536373539393535363232353137-eewp9wlm45rf > CSeq: 2 INVITE > Max-Forwards: 70 > User-Agent: snom320/8.7.5.44 > Contact: ;reg-id=1 > X-Serialnumber: 000XXX > P-Key-Flags: keys="3" > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Session-Expires: 600 > Min-SE: 90 > Authorization: Digest > username="200",realm="asterisk",nonce="7b2d56ec",uri="sip:*7...@sip.test.dk > ",response="7a9fe1f24a6f7585fb7323237a000167",algorithm=MD5 > Content-Type: application/sdp > Content-Length: 476 > > v=0 > o=root 558099897 558099897 IN IP4 172.16.1.29 > s=call > c=IN IP4 172.16.1.29 > t=0 0 > m=audio 60812 RTP/SAVP 9 8 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ysn2nTlXXAuZYcpOhf1g/h+oG > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > m=audio 60812 RTP/AVP 9 8 101 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=sendrecv > > > > > > - > --- > Alexey Vasilyev > -- > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double SDP
If you don't want to have both in the second INVITE, you can try putting both rtpengine_offer calls in branch routes instead. I haven't worked with rtpengine, but with other messages changes like this if you place them in the branch route then they affect only the current branch; after failure the original message will be returned and you may then be able to add RTP/SAVP only. Ben Newlin On 9/4/19, 8:27 AM, "Users on behalf of Alexey Vasilyev" wrote: This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This is Invite from snom phone, for example: Sent to tls:135.42.212.82:5061 at Sep 4 14:19:18.641 (1383 bytes): INVITE sip:*7...@sip.test.dk SIP/2.0 Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport From: "Demo" ;tag=ncsplp1nvz To: Call-ID: 313536373539393535363232353137-eewp9wlm45rf CSeq: 2 INVITE Max-Forwards: 70 User-Agent: snom320/8.7.5.44 Contact: ;reg-id=1 X-Serialnumber: 000XXX P-Key-Flags: keys="3" Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 600 Min-SE: 90 Authorization: Digest username="200",realm="asterisk",nonce="7b2d56ec",uri="sip:*7...@sip.test.dk",response="7a9fe1f24a6f7585fb7323237a000167",algorithm=MD5 Content-Type: application/sdp Content-Length: 476 v=0 o=root 558099897 558099897 IN IP4 172.16.1.29 s=call c=IN IP4 172.16.1.29 t=0 0 m=audio 60812 RTP/SAVP 9 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ysn2nTlXXAuZYcpOhf1g/h+oG a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=audio 60812 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv - --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double SDP
This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This is Invite from snom phone, for example: Sent to tls:135.42.212.82:5061 at Sep 4 14:19:18.641 (1383 bytes): INVITE sip:*7...@sip.test.dk SIP/2.0 Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport From: "Demo" ;tag=ncsplp1nvz To: Call-ID: 313536373539393535363232353137-eewp9wlm45rf CSeq: 2 INVITE Max-Forwards: 70 User-Agent: snom320/8.7.5.44 Contact: ;reg-id=1 X-Serialnumber: 000XXX P-Key-Flags: keys="3" Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 600 Min-SE: 90 Authorization: Digest username="200",realm="asterisk",nonce="7b2d56ec",uri="sip:*7...@sip.test.dk",response="7a9fe1f24a6f7585fb7323237a000167",algorithm=MD5 Content-Type: application/sdp Content-Length: 476 v=0 o=root 558099897 558099897 IN IP4 172.16.1.29 s=call c=IN IP4 172.16.1.29 t=0 0 m=audio 60812 RTP/SAVP 9 8 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ysn2nTlXXAuZYcpOhf1g/h+oG a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=audio 60812 RTP/AVP 9 8 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv - --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Double SDP
No it is either rtp/avp or rtp/savp.
On Wed, 4 Sep 2019, 13:54 Dragomir Haralambiev, wrote:
> Hello,
>
> I try setup Opensips to make local calls using device with encryption.
>
> In this example, it is not known UA 2 whether it supported (RTP/SAVP).
> UA 1 (RTP/AVP) > Opensips+Rtpengine -> UA 2
>
> 1. I try to make call with:
> rtpengine_offer("RTP/AVP replace-session-connection replace-origin
> ICE=remove");
>
> 2. If UA 2 is not supported RTP/AVP (use encryption always) return 488
> (Not Acceptable Media).
>
> failure_route[local] {
>
> if (t_check_status("488")) {
> rtpengine_delete();
> rtpengine_offer("RTP/SAVP replace-session-connection replace-origin
> ICE=remove");
> t_on_failure("local");
> t_relay();
> exit;
>};
> }
> This is INVITE from Opensips to UA 2.
> The Session Description Protocol contains two parts. One with RTP/AVP and
> the other with RTP/SAVP.
> Is this correct?
>
> INVITE
> .
> v=0
> o=- 13211994121466145 1 IN IP4 84.21.15.45
> s=X-Lite release 5.6.1 stamp 99142
> c=IN IP4 84.21.15.45
> t=0 0
> m=audio 51334 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=rtcp:51335
> v=0
> o=- 13211994121466145 1 IN IP4 84.21.15.45
> s=X-Lite release 5.6.1 stamp 99142
> c=IN IP4 84.21.15.45
> t=0 0
> m=audio 51334 RTP/SAVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=rtcp:51335
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:BQZW8OxjNdLM/Py8clP7nGLRPUXSDNTIhGS84YoN
> a=crypto:2 AES_CM_128_HMAC_SHA1_32
> inline:eC/i/z6ll3sZL/rhqfFcqK+q/atpInrBV+WBl921
> a=crypto:3 AES_192_CM_HMAC_SHA1_80
> inline:1Zz46W/M4YkZOWSebTcZwlZspR7t3n5e0NLx36DLREpyrFzYj7I
> a=crypto:4 AES_192_CM_HMAC_SHA1_32
> inline:43+DC+ZYBHrRWweVfJjP/U/3PFxd0QIyi5XHq7Oq8nnKS2w9lpQ
> a=crypto:5 AES_256_CM_HMAC_SHA1_80
> inline:HVjfOVa4qezhGqdKYOxD3KiZFllBUA6G7NCnnESuYHoy8Ha+pTQi57H0knlobg
> a=crypto:6 AES_256_CM_HMAC_SHA1_32
> inline:YbVeh5vrcQbMPKY9g13QCEBD7KZsin2wXWjQA+mEKfL0T8uMKE0+Dn2comPzmA
> a=crypto:7 F8_128_HMAC_SHA1_80
> inline:lLQrgIsJWoTb37NCXKJrB88aSBX3/ETWFyPSnOu1
> a=crypto:8 F8_128_HMAC_SHA1_32
> inline:+LuITZ9LDvcQUv2O7t9FBztNUOkHhgpmj91w2c6w
> a=crypto:9 NULL_HMAC_SHA1_80
> inline:y+3sqIOeZXDPG0mhYsz234s7Jrq3EOblQWT5bc0w
> a=crypto:10 NULL_HMAC_SHA1_32
> inline:6oMnVQvKFtJeQnHyyCY6ECDLwIjgn3kGyxYPm+nT
> a=setup:actpass
> a=fingerprint:sha-1
> A2:EF:11:86:E9:68:C9:8F:D6:86:33:07:BF:D1:6C:DD:6B:D8:FB:C3
>
>
>
> The problem is that I get a one-way voice.
>
> Best regards,
> Dragomir
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>
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[OpenSIPS-Users] Double SDP
Hello,
I try setup Opensips to make local calls using device with encryption.
In this example, it is not known UA 2 whether it supported (RTP/SAVP).
UA 1 (RTP/AVP) > Opensips+Rtpengine -> UA 2
1. I try to make call with:
rtpengine_offer("RTP/AVP replace-session-connection replace-origin
ICE=remove");
2. If UA 2 is not supported RTP/AVP (use encryption always) return 488 (Not
Acceptable Media).
failure_route[local] {
if (t_check_status("488")) {
rtpengine_delete();
rtpengine_offer("RTP/SAVP replace-session-connection replace-origin
ICE=remove");
t_on_failure("local");
t_relay();
exit;
};
}
This is INVITE from Opensips to UA 2.
The Session Description Protocol contains two parts. One with RTP/AVP and
the other with RTP/SAVP.
Is this correct?
INVITE
.
v=0
o=- 13211994121466145 1 IN IP4 84.21.15.45
s=X-Lite release 5.6.1 stamp 99142
c=IN IP4 84.21.15.45
t=0 0
m=audio 51334 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:51335
v=0
o=- 13211994121466145 1 IN IP4 84.21.15.45
s=X-Lite release 5.6.1 stamp 99142
c=IN IP4 84.21.15.45
t=0 0
m=audio 51334 RTP/SAVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:51335
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:BQZW8OxjNdLM/Py8clP7nGLRPUXSDNTIhGS84YoN
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:eC/i/z6ll3sZL/rhqfFcqK+q/atpInrBV+WBl921
a=crypto:3 AES_192_CM_HMAC_SHA1_80
inline:1Zz46W/M4YkZOWSebTcZwlZspR7t3n5e0NLx36DLREpyrFzYj7I
a=crypto:4 AES_192_CM_HMAC_SHA1_32
inline:43+DC+ZYBHrRWweVfJjP/U/3PFxd0QIyi5XHq7Oq8nnKS2w9lpQ
a=crypto:5 AES_256_CM_HMAC_SHA1_80
inline:HVjfOVa4qezhGqdKYOxD3KiZFllBUA6G7NCnnESuYHoy8Ha+pTQi57H0knlobg
a=crypto:6 AES_256_CM_HMAC_SHA1_32
inline:YbVeh5vrcQbMPKY9g13QCEBD7KZsin2wXWjQA+mEKfL0T8uMKE0+Dn2comPzmA
a=crypto:7 F8_128_HMAC_SHA1_80
inline:lLQrgIsJWoTb37NCXKJrB88aSBX3/ETWFyPSnOu1
a=crypto:8 F8_128_HMAC_SHA1_32
inline:+LuITZ9LDvcQUv2O7t9FBztNUOkHhgpmj91w2c6w
a=crypto:9 NULL_HMAC_SHA1_80 inline:y+3sqIOeZXDPG0mhYsz234s7Jrq3EOblQWT5bc0w
a=crypto:10 NULL_HMAC_SHA1_32
inline:6oMnVQvKFtJeQnHyyCY6ECDLwIjgn3kGyxYPm+nT
a=setup:actpass
a=fingerprint:sha-1
A2:EF:11:86:E9:68:C9:8F:D6:86:33:07:BF:D1:6C:DD:6B:D8:FB:C3
The problem is that I get a one-way voice.
Best regards,
Dragomir
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