Re: [OpenSIPS-Users] Siprec subscribe-answer rtpengine flags

Dienstag, 10. Juni 2025 09:58 An: users@lists.opensips.org Betreff: Re: [OpenSIPS-Users] Siprec subscribe-answer rtpengine flags Hi, Walter! You can pass the allow-transcoding flags in the $siprec(media) variable, before engaging recording: $siprec(media) = "allow-transcoding"; Best regards, Răzvan

Re: [OpenSIPS-Users] Siprec subscribe-answer rtpengine flags

n Răzvan Crainea Gesendet: Dienstag, 10. Juni 2025 09:58 An: users@lists.opensips.org Betreff: Re: [OpenSIPS-Users] Siprec subscribe-answer rtpengine flags Hi, Walter! You can pass the allow-transcoding flags in the $siprec(media) variable, before engaging recording: $siprec(media) = "allow-

Re: [OpenSIPS-Users] SIPREC start on 18x/SDP too early

Hi, Walter! The answer is actually a mixed of both - initially, the module started recording only on 200 OK - later on, there was a request to allow it on 183 as well, because it contains media too - this change has been made, but apparently docs were not updated (most likely my bad here, apol

Re: [OpenSIPS-Users] Siprec subscribe-answer rtpengine flags

Hi, Walter! You can pass the allow-transcoding flags in the $siprec(media) variable, before engaging recording: $siprec(media) = "allow-transcoding"; Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 6/3/25 11:50

[OpenSIPS-Users] SIPREC start on 18x/SDP too early

Hi! Acc to the docs https://opensips.org/docs/modules/3.5.x/siprec.html#func_siprec_start_recording the INVITE to the SRS should start on connect 200 OK. But it starts as soon as a SDP body is replied in the recorded call. E.g. in the 183/SDP or 180/SDP. Is this intensionally and the document

[OpenSIPS-Users] Siprec subscribe-answer rtpengine flags

Hi! I’ve successfully managed to record a call via SIPREC module using OpenSIPS 3.6.0 and RTPEngine 13.2. But I needed to do a trick to have the subscribe-answer to allow transcoding: I used a template „default“ in the rtpengine.conf to set [templates] default = allow-transcoding asymmetric T

Re: [OpenSIPS-Users] [SIPREC] Send re-INVITE for exising siprec session

Hi, Stefan! Unfortunately conference calls are not yet supported. Although the code has been designed with this idea in mind, the number of participants is hardcoded to 2, and there is no way to push more participants/calls into an existing SIPREC session. Best regards, Răzvan Crainea OpenS

[OpenSIPS-Users] [SIPREC] Send re-INVITE for exising siprec session

Hello, I've been trying the opensips siprec module (+rtpengine) with an SRS server and is working pretty nice so far. I have this setup in which multiple phones call into a conference and I want to ask if the following is implemented on the siprec module side: 1. First user calls into confe

[OpenSIPS-Users] SIPREC issue in opensips 3.4

Hello all, We are facing issues while configuring the SIPREC module. opensips through an error at Initial INVITE The errors in the log are as follows: ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established! ERROR:siprec:src_start_recording: could not start recording! ERROR:siprec:tm_

Re: [OpenSIPS-Users] siprec issue when listening on multiple sockets.

found it, use 2x force send socket. On 2/07/2021 11:44, johan wrote: > Hi, > > My server has 2 listening sockets. > > my phones register on 192.168.68.115 > > my siprec server is in 10.x.x.x segment. > > my ip config is like htis : > > 2: ens33: mtu 1500 qdisc pfifo_fast > state UP group default

[OpenSIPS-Users] siprec issue when listening on multiple sockets.

Hi, My server has 2 listening sockets. my phones register on 192.168.68.115 my siprec server is in 10.x.x.x segment. my ip config is like htis : 2: ens33: mtu 1500 qdisc pfifo_fast state UP group default qlen 1000     link/ether 00:0c:29:80:52:64 brd ff:ff:ff:ff:ff:ff     inet 192.168.68.115/

Re: [OpenSIPS-Users] SIPREC.

You are right.  Thank you for the suggestion. On 25/06/2021 12:04, Răzvan Crainea wrote: > I am 90% sure this has been fixed in the latest sources. The 10% > hesitation comes from the fact that you haven't specified the version > you are running. > > Best regards, > > Răzvan Crainea > OpenSIPS Cor

Re: [OpenSIPS-Users] SIPREC.

version: opensips 3.2.0-beta (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, CC_O0, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method

Re: [OpenSIPS-Users] SIPREC.

I am 90% sure this has been fixed in the latest sources. The 10% hesitation comes from the fact that you haven't specified the version you are running. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 6/25/21 12:52 PM, johan wrote: hmmnm, Core was ge

Re: [OpenSIPS-Users] SIPREC.

hmmnm, Core was generated by `/data/opensips/sbin/opensips -P /var/run/opensips/opensips.pid -m 64 -M 4 -u ro'. Program terminated with signal SIGSEGV, Segmentation fault. #0  0x7f1677a72fda in notification_listener_process (rank=0) at notification_process.c:345 345        *rtpp_notify_process

Re: [OpenSIPS-Users] SIPREC.

No, SIPREC can not currently work with rtpengine due to the fact that RTPEngine cannot fork the media stream to a new destination, at least not as RTP. AFAIK, only RTPProxy can do this now. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 6/25/21 11:44

[OpenSIPS-Users] SIPREC.

Hi, can the opensips configuration for siprec work with rtpengine ? To me it seems that it is dependant on rtpproxy.    So how can I combine siprec with rtpengine for a script in which rtpengine is always in the middle. wkr, OpenPGP_0xD7D896F7DDA70EC3.asc Description: OpenPGP public key O

Re: [OpenSIPS-Users] Siprec contact header

Hi Răzvan, Thank you for your feedback. I opened bug report #2383 on Github. With regards, Jofi Date: Fri, 22 Jan 2021 16:57:04 +0200 From: Răzvan Crainea To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Siprec contact header Message-ID: Content-Type: text/plain; charset=utf-8

Re: [OpenSIPS-Users] Siprec contact header

Hi, Jofi! Unfortunately this is not implemented in OpenSIPS right now. Can you please open a bug report for this? In the meantime, as a workaround, you can "catch" the INVITE in the local_route and modify the contact accordingly. Best regards, Răzvan Crainea OpenSIPS Core Developer http://ww

[OpenSIPS-Users] Siprec contact header

Hello All, I have a question regarding contact header generated by siprec module. According to rfc7866 (6.1.1): The SRC MUST include the "+sip.src" feature tag in the Contact URI, defined in this specification as an extension to [RFC3840 ], for all RSs

Re: [OpenSIPS-Users] siprec module does response re-invite from a siprec server.

Hello Razvan, Thanks for your prompt response. I opened a feature request. https://github.com/OpenSIPS/opensips/issues/2167 Thanks & Regards *Diptesh Patel* Software Developer Ecosmob Technologies Ltd, Ahmedabad Mo:*+919898962659* On Fri, Jul 10, 2020 at 9:05 PM Răzvan Crainea wrote: > Hi,

Re: [OpenSIPS-Users] siprec module does response re-invite from a siprec server.

Hi, Diptesh! Unfortunately reinvites from the SIPREC servers are not supported by the module right now. Please open a feature request for this. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 7/10/20 4:28 PM, Diptesh Patel wrote: Hello, I am using

[OpenSIPS-Users] siprec module does response re-invite from a siprec server.

Hello, I am using opensips 2.4.7 and opensips is working as a siprec client with rtpproxy. It is working fine but it does not respond to any re-invite from siprec server as a refresher after a few seconds and the siprec server rejects the call with BYE due to timeout and recording stops there. Is

Re: [OpenSIPS-Users] SIPREC server does not send BYE while capturing Early-Media.

Hi, Diptesh! This is indeed a limitation due to the fact that the module was initially designed to work only starting from a 200 OK. Please open a bug report for this on github so we can address it. Best regards, Răzvan On 6/10/20 7:39 AM, Diptesh Patel wrote: Hello, I am using the SIPREC

[OpenSIPS-Users] SIPREC server does not send BYE while capturing Early-Media.

Hello, I am using the SIPREC module to send the request to SRS with fail-over. While recording early media and after Originator Cancel OpenSIPS does not send a BYE request to the SRS. Please look at the script snippet of how I implemented that. if (is_method ("INVITE")) { if (siprec_st

Re: [OpenSIPS-Users] siprec with rtpproxy

Hi, Martin! You are right, the c= element is missing in the SDP. And according to the RFC[1], the body should have at least one c= element. Regarding the o= line, the IP over there is not that important, and it can be anything, provided that the entire o= line is unique. Now the main problem

[OpenSIPS-Users] siprec with rtpproxy

Hi, I am having issues with siprec and rtpproxy. Running the latest 2.4 dev version. The issue I am having is that when an 200 OK is received, the siprec is sending an incorrect SDP block to the recording provider. As a result of this they are unable to process the recording properly. For an ex