Re: [OpenSIPS-Users] codec stripping with rtpengine
Razvan, Thanks for the info I'll get the latest 2.3 and compile it. Ryan Esty Senior Software Engineer NEC Enterprise Communication Technologies, Inc. (Cheshire) 203-718-6268 From: Users <users-boun...@lists.opensips.org> on behalf of Răzvan Crainea <raz...@opensips.org> Sent: Monday, April 16, 2018 2:41 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine Hi, Ryan! That error is actually triggered by opensips, rtpengine module. There was a bug related to this issue that was fixed on 21st of February, and did not catch the 2.3.3 release. The latest 2.3 nightly has this fixed, so I'd suggest you to use that opensips version. Best regards, Răzvan On 04/13/2018 04:31 PM, Esty, Ryan wrote: > Razvan, > > Rtpengine is printing out this: rtpengine:parse_flags: error processing flag > `codec-strip-VP8': unknown error. As I look at it now I don't see how it > could be your issue. You aren't modifying the flag I sent to rtpengine. If it > helps my version of opensips is 2.3.3. > > Ryan > > > -Original Message- > From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan > Crainea > Sent: Friday, April 13, 2018 9:17 AM > To: users@lists.opensips.org > Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine > > Hi, Ryan! > > I think the issue you are talking about is related to OpenSIPS, rather than > rtpengine, since you are getting the error in OpenSIPS, is that right? > > Can you confirm what version of OpenSIPS you are using? > > Best regards, > Răzvan > > On 04/13/2018 03:42 PM, Esty, Ryan wrote: >> Bogdan-Andrei, >> >> Thanks for the information just in case someone else looks for this, >> this is the tracker https://github.com/sipwise/rtpengine/issues/525. >> >> Ryan >> >> *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org] >> *Sent:* Thursday, April 12, 2018 4:08 PM >> *To:* OpenSIPS users mailling list <users@lists.opensips.org>; Esty, >> Ryan <ryan.e...@necect.com> >> *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine >> >> Hi Ryan, >> >> yeah, this happens because OpenSIPS applies all the changes at the >> end, when the message is about to be sent out. As a side effect, when >> sending the SDP to rtpengine, opensips does not see its own previous >> changes over the body (changes are still pending). >> Usually there are easy workarounds for this, but in this case it looks >> like bug to me. Could you please open a bug report the the github tracker. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> >> http://www.opensips-solutions.com >> >> OpenSIPS Summit 2018 >> >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 04/09/2018 05:22 PM, Esty, Ryan wrote: >> >> Hi opensips list, >> >> First some background I’m trying to use opensips as a webrtc proxy. >> I found out that the packets for the invite going to my sip server >> are too big for my sip server. It doesn’t like packets to be over >> 4000 bytes. I’m trying to take what I can out of the sip packets >> like codes I know the other side can’t do. First codec stripping >> works but only with the audio codecs. If I try to strip a video >> codec the packet gets mangled. This is probably a bug in rtpengine >> and not opensips. I was hoping if anyone has any idea how I might >> get my invite packets smaller? The webrtc side is generating ssrc >> lines in my sdp. I’m trying to strip them out but I’m not sure if >> rtpengine is putting them back in or not. Before my rtpengine_offer >> I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has >> all the ssrc lines in it. >> >> Ryan >> >> >> >> >> ___ >> >> Users mailing list >> >> Users@lists.opensips.org <mailto: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 >> > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > __
Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi, Ryan! That error is actually triggered by opensips, rtpengine module. There was a bug related to this issue that was fixed on 21st of February, and did not catch the 2.3.3 release. The latest 2.3 nightly has this fixed, so I'd suggest you to use that opensips version. Best regards, Răzvan On 04/13/2018 04:31 PM, Esty, Ryan wrote: Razvan, Rtpengine is printing out this: rtpengine:parse_flags: error processing flag `codec-strip-VP8': unknown error. As I look at it now I don't see how it could be your issue. You aren't modifying the flag I sent to rtpengine. If it helps my version of opensips is 2.3.3. Ryan -Original Message- From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Friday, April 13, 2018 9:17 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine Hi, Ryan! I think the issue you are talking about is related to OpenSIPS, rather than rtpengine, since you are getting the error in OpenSIPS, is that right? Can you confirm what version of OpenSIPS you are using? Best regards, Răzvan On 04/13/2018 03:42 PM, Esty, Ryan wrote: Bogdan-Andrei, Thanks for the information just in case someone else looks for this, this is the tracker https://github.com/sipwise/rtpengine/issues/525. Ryan *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org] *Sent:* Thursday, April 12, 2018 4:08 PM *To:* OpenSIPS users mailling list <users@lists.opensips.org>; Esty, Ryan <ryan.e...@necect.com> *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine Hi Ryan, yeah, this happens because OpenSIPS applies all the changes at the end, when the message is about to be sent out. As a side effect, when sending the SDP to rtpengine, opensips does not see its own previous changes over the body (changes are still pending). Usually there are easy workarounds for this, but in this case it looks like bug to me. Could you please open a bug report the the github tracker. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 04/09/2018 05:22 PM, Esty, Ryan wrote: Hi opensips list, First some background I’m trying to use opensips as a webrtc proxy. I found out that the packets for the invite going to my sip server are too big for my sip server. It doesn’t like packets to be over 4000 bytes. I’m trying to take what I can out of the sip packets like codes I know the other side can’t do. First codec stripping works but only with the audio codecs. If I try to strip a video codec the packet gets mangled. This is probably a bug in rtpengine and not opensips. I was hoping if anyone has any idea how I might get my invite packets smaller? The webrtc side is generating ssrc lines in my sdp. I’m trying to strip them out but I’m not sure if rtpengine is putting them back in or not. Before my rtpengine_offer I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has all the ssrc lines in it. Ryan ___ Users mailing list Users@lists.opensips.org <mailto: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 -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam ___ 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 -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] codec stripping with rtpengine
Razvan, Rtpengine is printing out this: rtpengine:parse_flags: error processing flag `codec-strip-VP8': unknown error. As I look at it now I don't see how it could be your issue. You aren't modifying the flag I sent to rtpengine. If it helps my version of opensips is 2.3.3. Ryan -Original Message- From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan Crainea Sent: Friday, April 13, 2018 9:17 AM To: users@lists.opensips.org Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine Hi, Ryan! I think the issue you are talking about is related to OpenSIPS, rather than rtpengine, since you are getting the error in OpenSIPS, is that right? Can you confirm what version of OpenSIPS you are using? Best regards, Răzvan On 04/13/2018 03:42 PM, Esty, Ryan wrote: > Bogdan-Andrei, > > Thanks for the information just in case someone else looks for this, > this is the tracker https://github.com/sipwise/rtpengine/issues/525. > > Ryan > > *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org] > *Sent:* Thursday, April 12, 2018 4:08 PM > *To:* OpenSIPS users mailling list <users@lists.opensips.org>; Esty, > Ryan <ryan.e...@necect.com> > *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine > > Hi Ryan, > > yeah, this happens because OpenSIPS applies all the changes at the > end, when the message is about to be sent out. As a side effect, when > sending the SDP to rtpengine, opensips does not see its own previous > changes over the body (changes are still pending). > Usually there are easy workarounds for this, but in this case it looks > like bug to me. Could you please open a bug report the the github tracker. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > >http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > >http://www.opensips.org/events/Summit-2018Amsterdam > > On 04/09/2018 05:22 PM, Esty, Ryan wrote: > > Hi opensips list, > > First some background I’m trying to use opensips as a webrtc proxy. > I found out that the packets for the invite going to my sip server > are too big for my sip server. It doesn’t like packets to be over > 4000 bytes. I’m trying to take what I can out of the sip packets > like codes I know the other side can’t do. First codec stripping > works but only with the audio codecs. If I try to strip a video > codec the packet gets mangled. This is probably a bug in rtpengine > and not opensips. I was hoping if anyone has any idea how I might > get my invite packets smaller? The webrtc side is generating ssrc > lines in my sdp. I’m trying to strip them out but I’m not sure if > rtpengine is putting them back in or not. Before my rtpengine_offer > I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has > all the ssrc lines in it. > > Ryan > > > > > ___ > > Users mailing list > > Users@lists.opensips.org <mailto: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 > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam ___ 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] codec stripping with rtpengine
Hi, Ryan! I think the issue you are talking about is related to OpenSIPS, rather than rtpengine, since you are getting the error in OpenSIPS, is that right? Can you confirm what version of OpenSIPS you are using? Best regards, Răzvan On 04/13/2018 03:42 PM, Esty, Ryan wrote: Bogdan-Andrei, Thanks for the information just in case someone else looks for this, this is the tracker https://github.com/sipwise/rtpengine/issues/525. Ryan *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org] *Sent:* Thursday, April 12, 2018 4:08 PM *To:* OpenSIPS users mailling list <users@lists.opensips.org>; Esty, Ryan <ryan.e...@necect.com> *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine Hi Ryan, yeah, this happens because OpenSIPS applies all the changes at the end, when the message is about to be sent out. As a side effect, when sending the SDP to rtpengine, opensips does not see its own previous changes over the body (changes are still pending). Usually there are easy workarounds for this, but in this case it looks like bug to me. Could you please open a bug report the the github tracker. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 04/09/2018 05:22 PM, Esty, Ryan wrote: Hi opensips list, First some background I’m trying to use opensips as a webrtc proxy. I found out that the packets for the invite going to my sip server are too big for my sip server. It doesn’t like packets to be over 4000 bytes. I’m trying to take what I can out of the sip packets like codes I know the other side can’t do. First codec stripping works but only with the audio codecs. If I try to strip a video codec the packet gets mangled. This is probably a bug in rtpengine and not opensips. I was hoping if anyone has any idea how I might get my invite packets smaller? The webrtc side is generating ssrc lines in my sdp. I’m trying to strip them out but I’m not sure if rtpengine is putting them back in or not. Before my rtpengine_offer I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has all the ssrc lines in it. Ryan ___ Users mailing list Users@lists.opensips.org <mailto: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 -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] codec stripping with rtpengine
Bogdan-Andrei,
Thanks for the information just in case someone else looks for this, this is
the tracker https://github.com/sipwise/rtpengine/issues/525.
Ryan
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Thursday, April 12, 2018 4:08 PM
To: OpenSIPS users mailling list <users@lists.opensips.org>; Esty, Ryan
<ryan.e...@necect.com>
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine
Hi Ryan,
yeah, this happens because OpenSIPS applies all the changes at the end, when
the message is about to be sent out. As a side effect, when sending the SDP to
rtpengine, opensips does not see its own previous changes over the body
(changes are still pending).
Usually there are easy workarounds for this, but in this case it looks like bug
to me. Could you please open a bug report the the github tracker.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 04/09/2018 05:22 PM, Esty, Ryan wrote:
Hi opensips list,
First some background I'm trying to use opensips as a webrtc proxy. I found out
that the packets for the invite going to my sip server are too big for my sip
server. It doesn't like packets to be over 4000 bytes. I'm trying to take what
I can out of the sip packets like codes I know the other side can't do. First
codec stripping works but only with the audio codecs. If I try to strip a video
codec the packet gets mangled. This is probably a bug in rtpengine and not
opensips. I was hoping if anyone has any idea how I might get my invite packets
smaller? The webrtc side is generating ssrc lines in my sdp. I'm trying to
strip them out but I'm not sure if rtpengine is putting them back in or not.
Before my rtpengine_offer I do a replace_body_all("a=ssrc.*\r\n," "") but the
invite still has all the ssrc lines in it.
Ryan
___
Users mailing list
Users@lists.opensips.org<mailto: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] codec stripping with rtpengine
Hi Ryan, yeah, this happens because OpenSIPS applies all the changes at the end, when the message is about to be sent out. As a side effect, when sending the SDP to rtpengine, opensips does not see its own previous changes over the body (changes are still pending). Usually there are easy workarounds for this, but in this case it looks like bug to me. Could you please open a bug report the the github tracker. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 04/09/2018 05:22 PM, Esty, Ryan wrote: Hi opensips list, First some background I’m trying to use opensips as a webrtc proxy. I found out that the packets for the invite going to my sip server are too big for my sip server. It doesn’t like packets to be over 4000 bytes. I’m trying to take what I can out of the sip packets like codes I know the other side can’t do. First codec stripping works but only with the audio codecs. If I try to strip a video codec the packet gets mangled. This is probably a bug in rtpengine and not opensips. I was hoping if anyone has any idea how I might get my invite packets smaller? The webrtc side is generating ssrc lines in my sdp. I’m trying to strip them out but I’m not sure if rtpengine is putting them back in or not. Before my rtpengine_offer I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has all the ssrc lines in it. Ryan ___ 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
